Principles of Public Address and Voice Evacuation

Content

1 Introduction 3 4. Microphones 17

2 Basics of sound 6 4.1 Common categories 17

2.1 Speech 8 4.2 Special microphones 18

2.2 Music 8 4.2.1 Evacuation Call Stations 18

2.3 Sound 9 4.2.2 Noise cancelling microphone 19

2.3.1 Ear-characteristics 9 4.2.3 Radio (Wireless) microphone 19

2.3.3 Sound Pressure Level 11 4.3 Microphone Technique 19

2.4 Sound propagation in air 11 5 Pre amplifiers and Amplifiers 21

3 Decibel (dB) 13 5.1 The pre-amplifier 21

3.1 Logarithmic characteristics of the ear 13 5.1.1 Equalizers 21

3.2 Power ratios 14 5.1.2 Basic tone controls 22

3.3 Voltage ratios 14 5.1.3 Equalizer types 22

3.4 dB references 15 5.2 The power amplifier 24

3.5 Calculations 16 5.2.1 Frequency Response 24

5.2.2 Power bandwidth 24

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Principles of Public Address and Voice Evacuation
5.2.3 Linear distortion 25 7.2.8 Technical Principles 36

5.2.4 Non linear distortion or clipping (THD) 25 8 Equalization and Acoustic Feedback 41

5.2.5 Rated Output Power 25 9 Time Delay 44

5.2.6 Temperature Limited Output Power (TLOP) 10 Automatic Volume Control 45

26
11 The Acoustic Environment 46
5.3 Voice Evacuation Amplifiers 26
11.1 Outdoors 46
5.4 Amplifier / loudspeaker interface 27
11.2 Indoors 49
6. Cable lengths. 29
11.2.1 Direct Field and reverberant Field 49
7. Speakers 30
11.3 Speech intelligibility graph STI & RASTI 52
7.1 Theory of operation 30
11.3.1 Articulation Losses of consonants in
7.2 Loudspeaker types 30 speech. (% ALcons) 53

7.2.1 Standard loudspeaker cabinets 30 11.3.2 Speech Transmission Index (STI & RASTI)
54
7.2.2 Ceiling loudspeakers 31
11.3.3 Subjective %ALcons and RASTI
7.2.3 Sound columns 32 requirements. 54
7.2.4 Horn loudspeakers 32 11.3.4 Converting RASTI to %ALcons 55
7.2.5 Full range high power loudspeakers 33 12. Loudspeaker placement and coverage56

7.2.6 EVAC Loudspeakers 33 12.1 Loudspeaker design steps 57

7.2.7 IP rating 35

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Principles of Public Address and Voice Evacuation

1 Introduction

Public Address and Voice Evacuation sound systems are used for amplification of speech or music to
enhance intelligibility or loudness by electro-acoustic means. The audience is served with a higher
degree of listening comfort and safe evacuation of audience and staff during an emergency situation is
ensured.

A public address and voice evacuation system Sound system design is a comprehensive
is designed primarily to carry live and recorded subject combining a chain of devices:
messages, signal tones and background music microphone, sound processing equipment,
(if required), from several different sources, to amplifier and loudspeaker, together with the
a number of selectable remote areas. acoustic environment, into a single system.

The microphone converts the acoustical The amplifier increases the level of the signal
vibrations, caused by an audio source, into an to one adequate for driving loudspeakers.
electrical signal.
The loudspeakers convert the electrical signal
The processing equipment modifies the back into vibrations, which are greatly
signal to compensate for deficiencies in the influenced by the acoustic environment, and in
source or environment. turn received by the ear of the listener.

This manual is intended to give readers with a technical background a reference to the various
aspects of audio engineering, sound system design and voice evacuation.

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Vertical Markets

Common applications where

Public Address and Public Address
& Voice Evacuation systems are
used would be: hotels, restaurants,
railway stations, airports, factories,
oil platforms, office buildings,
schools, shopping areas, ships
and exhibition areas.

Public address and Public Address & Voice Evacuation

A public address distribution system is designed primarily to carry live and recorded messages, signal
tones and background music (if required), from several different sources, to a number of selectable
remote areas, based on 100 Volts technology.

The public address system provides several • Calls via limiter,

functionalities for earlier mentioned
• fixed volume,
applications:
• priority handling,
• attention signals,
• live and pre-recorded calls, background
music (BGM),
• routing switches / zone selection,
• hidden controls and
• automatic volume control

Another important function of the public address system is mono sound:

→ With mono sound all the audio signals Mono systems can still be full-
are mixed and routed through the bandwidth and the big advantage to
same audio channel. The key is that mono is that everyone hears the very
the signal contains no level and arrival same signal. This makes mono sound
time/phase information that would suited for public address as they can
replicate or simulate directional cues. provide excellent speech intelligibility.

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Principles of Public Address and Voice Evacuation

The Public Address & Voice Evacuation When the term Voice Evacuation is added the
system basically provides the same system will have to comply with national or
functionality as the Public Address system. A international standards. In most of Europe the
Public Address & Voice Evacuation System EN 60849 and EN 54 (parts 16, 19 and 24)
however is apart from public announcements, determine the requirements for a Voice
scheduled events and background music Evacuation system. Outside Europe ISO 7240
provision also used to bring people to safety in (parts 16 and 19), UL864 and UL1711 are
emergency situations. applicable a.o.

As Public Address & Voice Evacuation system is expected to operate at all times and at any place, the
correct operation is guaranteed by continuous monitoring, continuous control and indicating functions.

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2 Basics of sound

Sound waves are generated by air particles reaction with other particles next to them. In
being set in motion by physical movement this way, a movement of air is transmitted in
(such as the bow being drawn across a violin all directions, by expanding and compressing
or a vibrating cone of a loudspeaker). Once air. So sound is energy that is transmitted by
the particles start moving, they begin a chain pressure waves in air.

We can make a distinction in the waveforms:

→ Spherical waves originated from an in

all directions radiating point source.

→ Cylindrical waves originated from a line

source.

→ Plane waves originated from a plane

source.

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Principles of Public Address and Voice Evacuation

On relative long distances from the sound in above picture where the sound is moving in
source (r >> source dimensions >> jumps of e.g. 1m. The total sound power (W)
wavelength), we apply generally the spherical is in all cases the same but the sound
wave attenuation r2 but regard the waveform pressure (N/m2) or sound intensity (W/m2) is
as a plane wave. The justification can be seen decreasing with r2.

Sound, as we refer to in this manual, Dependent on our terms of reference dynamic

generally consists of speech, music, alarm range has different meanings:
signals or attention tones. Any kind of
transmission or registration of sound, when 1. In acoustics, the quietest sound level to the
converted into electrical signals, imposes a loudest one.
limitation on the dynamic range, frequency
2. In music, the difference between
response, intelligibility and natural quality. The
pianissimo and fortissimo.
limitation of the dynamic range is the most
prominent and the most important. 3. In sound engineering we prefer to express
it as the difference between the maximum
incidental peak value and the minimum
value of the converted electrical signal.

With different measuring instruments (oscilloscope, level meter, VU-meter, etc.) we can analyze the
signal, but dependent on the applied instrument’s characteristics (integration time) different levels will
be shown.

Oscilloscope with memory

Sparks = Integral time 0 ms =
screen
Peak level meter (e.g. on
Peak = Integral time < 5 ms =
mixing desks)
Fast
= Integral time 125 ms = Sound level meter (SLM)
(Short Time Average)
Volume Unit meter (e.g. for
VU = Integral time 270 ms =
amplifier)
Slow = Integral time 4 s = Sound level meter (SLM)
LTA Used for Amplifier cooling
= Integral time 30 s =
(Long Time Average) design

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2.1 Speech

Speech consists of words and

pauses. Words contain both
vowels and consonants. Speech
has loudness variation and
frequency variation. Dependent
on the voice strength the
frequency spectrum (which is the
lowest bass sound through to the
highest treble one) is changing
according the diagram. The lines
in the graph represent the
average level per 1/3 octave.

Let's look at loudness first. The vowels in a directional. Our principle aim is to deliver this
sentence have a frequency spectrum below complete speech spectrum to the listener’s
1000Hz, and they create the impression of ears as unchanged as possible.
loudness. The human mouth producing these Unfortunately various acoustical phenomena,
sounds does so with a wide opening angle which are discussed throughout this book,
and in indoor situations can hit hard surfaces play their part in altering the speech
within range like walls and ceiling etc, so spectrum, at times making it impossible for
easily causing reverberation. listeners to understand what is being said.

The consonants of the words in a sentence, Because of this certain techniques are
having a frequency spectrum above 1000 Hz, employed to compensate for these
provide the articulation. The human mouth phenomena in order to make the speech
produces these sounds with a narrow intelligible.
opening angle and, because of this, is rather

2.2 Music

As with any sound transmission, the two frequencies, the music will lack the depth to
most prominent features of music reproduce bass instruments fully. If it is
reproduction are the dynamic range and the limited at the higher frequencies, harmonics,
frequency response. If the dynamic range is which are vital for instrument recognition, will
limited, the music will appear emotionally flat, not be fully present, causing the music to
lacking both subtlety and excitement. If the sound dull.
frequency response is limited at the lower

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2.3 Sound

Sound is a series of vibrations compressing The threshold of hearing at 1 kHz is: 20

2 -5
and rarefying the air. Loudness is the µN/m = 20 µPa = 2 x 10 Pa. (Pa = Pascal =
2
subjective experience of sound level. Since, N/m ). Sound Pressure related to this
when we measure sound, we refer to reference level is expressed in dB (SPL).
changes in air pressure, a reference related
to pressure must be used. As the sound pressure level is increased, a
point is finally reached, just short of being
The reference used is the level of sound at 1 painful to the ear, called the 'threshold of
kHz, which is barely perceptible to people pain', which, using the 1 kHz reference
with normal hearing, being the quietest sound frequency, corresponds to 20 Pa.
pressure that an average person can hear.
This is called the 'threshold of hearing'. Since the 0 dB (SPL) absolute reference is
20 µPa;

2.3.1 Ear-characteristics

As is shown in the graph, the Sound

Pressure Level (SPL) at the
threshold of hearing varies with
frequency. Because of this it would
require 60 dB (SPL) at 30 Hz to
produce the same impression of
loudness as 0 dB (SPL) at 1 kHz.
SPL

The threshold of hearing represents

the bottom limit of a series of ‘equal
loudness’ contours, which are also
shown.

In studying the graph, we notice two important factors:

→ Firstly, very much more energy is impression which corresponds with the
needed to produce a bass signal of a mirror image of the 20 dB equal
given loudness, when compared with a loudness contour. If the noise level is
2 or 3 kHz signal - an important now raised a different impression of
consideration in any system used to the same noise is received. This is due
reproduce music. to the ear responding to the noise
according to the changing curves of
→ The second point is that if noise, equal loudness. In other words, all
having a broad frequency spectrum, frequencies are present in the signal
with a level of, say, 20 dB (SPL) is but depending on the level, they will be
reproduced the listener will have an heard in different relationships.

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2.3.2 Weighting

In order to imitate this characteristic of the ear, a sound level meter often incorporates different
filter curves which correspond with this subjective hearing. There are three types of curves
internationally standardized and they are called A-, B-, and C- weighting.

A-curve B-curve C-curve

→ Follows the frequency → Follows the frequency → Follows the frequency

sensitivity of the human sensitivity of the human sensitivity of the human
ear at low levels. This is ear at moderate levels, ear at very high noise
the most commonly used used in the past for levels. The C-weighting
weighting scale and predicting performance scale is quite flat, and
should theoretically be of loudspeakers and therefore includes much
used only for stereos, but not industrial more of the low-
measurements below 40 noise. frequency range of
dB (SPL). Many simple sounds than the A and B
sound level meters scales.
though are equipped with
an A-curve filter and
nowadays the majority of
acoustic measurements
are taken solely with A-
weighting. Noise
measurements made
with the A-weighting
scale are designated
dBA (SPL).

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2.3.3 Sound Pressure Level

The accompanying chart shows the sound pressures in dB(SPL), for several common sound
sources.

2.4 Sound propagation in air

Sound could most simply be defined as a Important physical parameters, which influence
series of vibrations compressing and the propagation of sound in air are:
thinning the air. To be transmitted, sound
relies on a vibrating object (vocal chords, • f = frequency (Hz)
loudspeaker, breaking window, etc.) which
• v = velocity (m/s)
imparts its motion to surrounding molecules
or particles.
• λ = wavelength (m)

• p = pressure (Pa)

• T = temperature (K)

Velocity of sound

The velocity of sound is determined mainly by the temperature. For normal conditions, in air, the
velocity may be calculated by:

At 20°C: v = 20 √ T T = temperature in Kelvin (0°C = 273 K)

v = 20 √ 293

v = 342,3 ≅ 340 m/s

The relationship between frequency, Using these equations, it is seen that at 1 kHz
wavelength and velocity is given by: λ = v/f at 20°C, the wavelength is

λ = 340 / 1000 = 0,340 m

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Air Absorption

Listening to sound on distance makes us Because the humidity affects the amount of
aware of frequency dependant attenuation water molecules in the air, also the
due to air-absorption, the higher the attenuation of a sound signal is affected. This
frequency the more attenuation. This means that a relative humidity (RH) of 20%
attenuation for a frequency of 500 Hz equals attenuates a 4 kHz signal by 0,09 dB/meter,
0.3 dB per 100 meter, for 2000 Hz equals 1 whilst a relative humidity of 80% attenuates a
dB per 100 meter and for 8000 Hz equals 7 4 kHz signal by 0.02 dB/meter. The humidity
dB per 100 meter (RH=70%). level should certainly not be discounted since
its effect can be quite dramatic.

Reverberation time

The effect of reverberation time (RT60) in a

room with volume (V) and surface (S):
with: α = average absorption coefficient
RT60 = 0,161 V/(αS + 4mV) -1
m = attenuation constant (m )

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3 Decibel (dB)

The use of the decibel (dB) notation system is common in sound and communications work. This
system allows meaningful scale compression or expansion as required and greatly simplifies
computations involving large quantities. Our human senses - touch, sight, hearing, sense of weight,
etc. - all function logarithmically. That is, in the presence of a stimulus the least perceptible change is
proportional to the already existing stimulus (Weber-Fechner law).

3.1 Logarithmic characteristics of the ear

To evaluate the ear’s behavior in respect to sensitivity for level differences, we can experiment as
follows:

The diagram shows two identical amplifiers noticeable difference will be heard until it
and loudspeakers with a signal generator receives 26% more power (26% of 126 mW =
switched to one then to the other alternately. 32 mW), which brings the higher loudspeaker
Initially the same power is supplied to each output to 126 + 32 = 158 mW. In this way, the
loudspeaker, e.g. 100 mW, and because of noticeable increase in loudness is obtained
this both their signals are of equal loudness. by raising the level in a given ratio, not by
As the power to one of the loudspeakers is adding specific amounts of power. Increasing
slightly increased no difference in loudness power in ten stages of 26% brings it to ten
will be heard, whilst continuing to listen to times its original level. This is a logarithm
them one at a time. Only when one increase not a linear increase.
loudspeaker receives 26% more power it will
sound noticeable louder. At this point e.g. A power increase of a factor 10 is one Bel,
126 mW is being fed to one loudspeaker and with each power increase of 26% being one
100 mW to the other. tenth of a Bel and called a decibel (dB). It
must be appreciated that the dB is only a
If the power of the other loudspeaker is also ratio, and that the ear hears the same
increased to 126 mW, the intensities will difference between 1 W and 2 W as between
again be equal. If the power to the first 100 W and 200 W.
loudspeaker is once again increased, no

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3.2 Power ratios

The Bel is defined as: Log P1/P2 , so the decibel (dB) is defined as: 10 Log P1/P2. The
amplification power ratio, expressed in dB, is given in the accompanying table. This shows, for
instance, that 3 dB amplification doubles the power, and that a 100 times increase in power gives
20 dB amplification.

3.3 Voltage ratios

• When 10V is connected to a 10 Ω resistor:

I = U/R = 1A

Power dissipated (P) = U x I = 10 W.

• When the voltage is doubled and still connected to the 10Ω resistor:

I = U / R = 2A

P = U x I = 40W, i.e. 4 times increase in power.

This shows that, in this case, doubling the voltage results in a quadrupled power; or to put it
another way, a doubling of the power (3 dB increase) will not result in a doubling of voltage. Since
power is dissipated in the same resistor:

Because 10 Log power ratio = 20 Log voltage be seen that an amplifier (or attenuator)
ratio, a gain of 3 dB gives a 2 x power gain, having a particular gain expressed in dB has
but only a 1.4 x voltage gain. In the same a different multiplicative effect dependent on
way, a 6 dB gain results in a 4 x power gain whether power gain or voltage gain is being
but only a 2 x voltage gain. From this it can considered.

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3.4 dB references

Though the decibel is only a ratio, it can be used to express absolute values if there is a given
reference. If, for example, a reference of 1 W is chosen, then 3 dB corresponds to 2 W, and 6 dB to
4 W and so on.

dBm - dBu

One of the common references used in the In practice however the resistance value is
past, due to its frequent application in frequently ignored when dBm is quoted and
Telecommunications is 1 milliwatt (mW) the reference is 775 mV only, this makes
across 600 Ohms, expressed as "dBm". (1 this reference incorrect. In fact the dBu is
mW across 600 ohms = 775 mV.) referred to the 775 mV regardless of the
impedance and is still commonly in use in
studio engineering.

dBV

This is the favourite and common reference

for electrical engineering. The reference is
1 Volt regardless of the impedance.
Corresponding dB values are measured in
dBV (e.g. 20 dBV = 10V).

dB(SPL)

There is another reference, which is used in to this reference level is expressed in dB

the measurement of sound pressure levels. (SPL). As the sound pressure level is
As we know, sound is basically a series of increased, a point is finally reached, just
vibrations compressing and rarefying the short of being painful to the ear, called the
air. Since, when we measure sound, we 'threshold of pain', which, using the 1 kHz
refer to changes in air pressure, a reference reference frequency, corresponds to 20 Pa.
related to pressure (the Sound Pressure Since the 0 dB (SPL) absolute reference is
Level) must be used. The reference used is 20 µPa;
the level of sound which is barely
perceptible to people with normal hearing,
being the quietest sound pressure that an
average person can hear at 1 kHz. This is
called the 'threshold of hearing'. At this Other important levels are:
point the threshold of hearing is very low:
20 µN/m2 = 20 µPa = 2 x 10-5 Pa. (Pa =
Pascal = N/m2) So Sound Pressure related

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3.5 Calculations

• When adding two unrelated sound sources, only their intensities (energy) should be added
together.

• Two different noise sources both producing 90 dB (SPL) would be experienced as:

• When subtracting two unrelated sound sources, only their intensities (energy) should be
subtracted:

The following graph shows how

to add or subtract levels in dB's
for non-related signals.

To add levels of non-related signals.

Enter the chart using the numerical difference value to the larger signal level to determine
between the two signal levels being added the total.
(top right of chart). Follow the line
corresponding to this value until it meets the Example: Combine a 75 dB signal with one of
curved line, then move left. The figure shown 80 dB. The difference between these figures
on the vertical scale at the left of the chart is is 5 dB. The 5 dB line intersects the curved
the numerical difference between the total line at 1.2 dB on the vertical scale. This
and larger of the two signal levels. Add this means that the total value is 80 + 1.2, or 81.2
dB.

To subtract levels of non-related signals.

If the numerical difference between the total The figure shown on the vertical scale at the
and the smaller of the two levels is between 3 left of the chart is the numerical difference
and 14 dB, enter the chart from the bottom. between the total and the unknown (the
Using the numerical difference, follow the line larger) level. Subtract this value from the total
corresponding to this value until it intersects to determine the unknown level.
the curved line, then follow the line to the left.

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4. Microphones

Microphones are a key factor in achieving The electrical current generated by a

high quality recordings and sound microphone is very small. Referred to as mic
reproduction. Having access to top-notch level, this signal is typically measured in milli
equipment helps, but more importantly, some volts. Before it can be used for anything
knowledge about mic basics and a few serious the signal needs to be amplified,
simple tips and tricks can make a substantial usually to line level (typically 0.5 -2V). Being
difference on your next project. a stronger and more robust signal, line level
is the standard signal strength used by audio
processing equipment and common domestic
equipment.

4.1 Common categories

There are two common categories of mics used in professional audio.

• Dynamic types operate when sound

waves strike a diaphragm attached to
a coil of wire. When the coil moves
within the magnetic structure of the
microphone, this creates an output
voltage. The process is exactly the
reverse of the way a speaker operates.
One variation of the dynamic approach
is the ribbon mic, which uses a thin
ribbon of metal that is placed between
the poles of a magnet. Note that most
ribbon mics are bidirectional, meaning
they pick sounds equally well from
either side of the mic.

• Condenser mics use an electrically-

charged, metallised diaphragm, placed
very close to a conductive back plate
and separated by a thin air layer.
Sound waves striking the diaphragm
cause a very small voltage change,
which is increased by a tiny amplifier
circuit within the mic body. As power is
required by both the mic capsule and
the amplifier, condenser mics must
have a power source, which can be a
battery inside the mic body or
"phantom" power coming from either
the mixing console or an external
power supply.

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4.2 Special microphones

A large number of special microphones are available, ranging from broadcast, through to the
individual requirements of different musical instruments. In the field of sound reinforcement and
public address there are again several different types of microphone likely to be encountered for
specialist applications.

4.2.1 Evacuation Call Stations

A voice evacuation system may have provision for emergency

microphone(s). In this case the emergency microphone is
equipped with:

a) priority over all inputs, including pre-recorded messages

b) an emergency microphone control to open the microphone channel

c) where a pre-announcement attention drawing signal is provided, an indicator adjacent to the

microphone shall show when the signal has finished and live speech can commence

d) when the emergency microphone control is operated, any audible indication that might
interfere with the use of the microphone shall be automatically muted;

e) Where more than one evacuation microphones are connected, they shall be configurable for
priority and only one emergency microphone shall be active at any one time.

Furthermore the emergency microphone capsule shall be supervised so to be continuously

informed about eventual short circuit in a transmission path to an emergency microphone
capsule.

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4.2.2 Noise cancelling microphone

This is essentially a hyper-cardioids the mouth, so filters have been built in to

microphone having an optimum speech ensure that the frequency response is flat
characteristic, and is designed for extremely when the sound source is close to the
noisy environments such as touring buses, microphone, and also that the bass content
factories, and supermarket floors. This type of the random noise is reduced.
of microphone must be held very close to

4.2.3 Radio (Wireless) microphone

Great freedom of movement is provided for connected to a small pocket transmitter,

the microphone user by the use of a allowing full hands-free use.
transmitter/receiver system. A FM signal
provides a link between either a hand-held When two or more radio (wireless)
or lavalier/lapel microphone and a receiver microphones are used in the same location,
connected to the sound system input. The care should be taken to ensure that they
hand held microphone has a built-in each operate on a different transmission
transmitter, while the lavalier model is frequency, otherwise conflicts will occur.

4.3 Microphone Technique

Microphones in the Bosch product range, are and back of the microphone are also picked
of advanced design, are very sensitive, and up, but at a much reduced level. This
reproduce the human voice with great clarity. characteristic gives them a high front to
Many of these microphones have a hyper random response index. Due to the fact that
cardioids response pattern, being particularly they are so directional, hyper cardioids
sensitive to sounds, which are generated at microphones operate particularly well in
the front, and on axis with the microphone difficult acoustic environments and in areas
body. Other sounds, generated at the sides with high background noise.

In order to optimize these or any microphone, it is important to be aware of certain operating

techniques:

→ The microphone should be pointing directly at, but placed a little below, the speaker's mouth.
This is to pick-up full spectrum sound including high frequencies and avoiding air blowing
frontal on the microphones diaphragm and causing “plops”.

→ The best distance from which to speak into a microphone is approximately 15 to 40

centimeters. If that distance is reduced greatly, a phenomenon, especially common to
(hyper)cardioids microphones, known as 'proximity effect' will occur. This is a very noticeable
increase in the bass content of the signal, making the voice muffled, and at times
unintelligible.

→ Speak at a consistent volume level.

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→ If the operator were to speak from a much greater distance than that recommended, the
microphone would also pick up other sounds in the room, effecting the overall clarity. This is
particularly unfortunate when the microphone is in the same room as the loudspeakers, due to
the fact that the amplified signal could be picked up by the microphone and amplified again. If
the amplification in this loop is allowed to continue, the disturbing phenomenon known as
acoustic feedback, or 'howl around', will occur.

→ If feedback does occur, do not cover the microphone with your hand; this makes the situation
worse. If you are very close to the microphone, moving backwards sometimes helps eliminate
feedback. The operator should then reduce the amplifier volume slightly, or use a tone control
or equalizer to attenuate the offending frequency somewhat.

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5 Pre amplifiers and Amplifiers

The public address amplifier exists of two sections, being the pre-amplifier (voltage gain) and the
output amplifier (power gain).

The pre-amplifier matches and amplifies the The power amplifier is used to amplify the
outputs of microphones, line inputs (CD, output power of a pre-amplifier and/or
tuners, PC) to provide a voltage level suitable distribution system to a level (100V in a
for driving the power amplifier. The pre- public address and voice evacuation system)
amplifier also normally incorporates the tone that will feed the loudspeakers properly.
controls, input sensitivity adjustments, and
master volume controls.

5.1 The pre-amplifier

The pre-amplifier is normally used for matching and amplifying small voltages, to provide a voltage
level, usually 500 mV or 1 V, which is suitable for driving the power amplifier.

Typical inputs to the pre-amplifier may be:

• moving coil (dynamic) microphone - 0,25 mV;

• condenser microphone - 3 mV;
• dynamic pick-up - 5 mV;
• domestic source (tuner, PC, CD etc.) - 250 mV;

From this range of input requirements two

inputs are often chosen: a microphone input
with a sensitivity of 0,5 mV to 1.5 mV; and a
music input of 100 mV to 1,5 V.

Tone controls, input sensitivity adjustments,

and master volume controls are usually
built into the pre-amplifier.

5.1.1 Equalizers

An Equalizer gives extensive control over Used with care, this can guarantee
the whole audio frequency spectrum by maximum amplification for the whole
means of gain and attenuation filters and frequency spectrum, at the same time
can be used for optimizing the frequency fighting the problem of acoustic feedback by
response of the sound system. reducing the level of frequencies which
cause it.

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5.1.2 Basic tone controls

→ If the bass or treble potentiometer is turned to the right, from its 0 ('flat') position, the gain is
increased, and the frequencies within its band of influence are amplified, giving an increase in
volume of the respective bass or treble frequencies. The 'lifting' of the treble frequencies is
particularly useful when it is desired to give speech greater clarity; helping it to 'cut through'
noisy environments.

→ If the potentiometer is turned to the left; the respective bass or treble frequencies are
attenuated. Bass attenuation is particularly useful in large rooms, where long reverberation
times at low frequencies cause problems.

→ Bass lift and treble attenuation is rarely required. Bass lift could be used when amplifying
music in a heavily damped room, where the bass frequencies would require reinforcement to
give the music more depth. Care should be taken though not to overload the loudspeakers
when amplifying the bass content of a signal.

5.1.3 Equalizer types

Band-pass filters

Bass and treble "Hi-Pass" and "Lo-Pass" (or attenuate all signals below or above a fixed
"cut-off") filters are intended to restrict the (normally very low or very high) frequency.
frequency band. Their purpose is to severely

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Parametric equalizer

A parametric equalizer is a unit with 3 or 4

filters, and the possibility to adjust the
frequency to be processed. The processing
consists of gain correction (+ , - ),and a
selection of the width (Q) of the frequency
band. This makes it possible to alter, if
necessary, a very small frequency band,
without affecting the neighboring
frequencies.

Because only a few filters are used, the

overall response tends to be quite smooth.

Parametric triple Q-filter

Basically a parametric equalizer but with

pre-set fixed (speech) centre frequencies
e.g. 1-2-4 kHz. This filter allows the operator
to select the width & slope of the frequency
band (Q) and presence or absence (Gain).
The unit is ideal for optimizing the
amplification of that part of the frequency-
band that is responsible for speech
intelligibility, it adds clarity and compensates
for air absorption. An adjustable bass cut
filter provides smooth roll off of the bass
content in the signal caused by e.g.
speaking too close to a cardioids
microphone.

Graphic equalizer

A Fixed Frequency or "Graphic" Equalizer example, one control at full attenuation and
often consists of 30 individual filter sections. its neighbor at full amplification.
Each control, which is often a sliding
potentiometer or "fader", effects a narrow The maximum level of both speech and
frequency band (third octave). The "peak", music is in the 250-500 Hz frequency range.
or maximum effect is at the centre of each The level at these frequencies should be
band, with the surrounding frequencies kept as near to 0 dB as possible to avoid
being effected to a proportionately lesser distortion due to a general level increase.
degree. The total frequency spectrum is
covered, allowing the signal to be sculptured
at several specific frequency bands as
desired. To avoid excessive phase shifting,
care should be taken to avoid extremes of
variation between adjacent controls with, for

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Principles of Public Address and Voice Evacuation

5.2 The power amplifier

The power amplifier is used to amplify the

output voltage of the pre-amplifier, distribution
system, or mixing desk, to a level that will feed
the loudspeakers properly. Depending on the
design philosophy of the manufacturers, the
input required to feed the amplifier at nominal
full power can range from 100 mV to 10 V.

Amplifiers used in public address and voice

evacuation systems use the 100 Volt line
principle. This type of amplifier is favorable if
long loudspeaker distances are involved. (This
principle is discussed in the following section)

5.2.1 Frequency Response

The above figure shows an amplifiers’ When this specification relates to power
frequency response. Specification states the amplifiers the level at which it is measured
frequencies at the points where the curves should be 10 dB below the rated output power.
have dropped by 3 dB. In the example, the Specifications should be read carefully. If a
frequency response is from 63 Hz to 16 kHz. manufacturer chooses -6 dB points as
reference, he is able to quote a frequency
response range which extends much wider
than more ethical competitors.

5.2.2 Power bandwidth

The Power bandwidth is the frequency distortion level (THD) as stated by the
range in which the amplifier can deliver its manufacturer (0.5% for PA amplifiers).
rated power (-3dB) with a maximum

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5.2.3 Linear distortion

If an amplifier is not capable of amplifying This unwanted modification of the signal is

the full frequency spectrum equally, the called linear distortion, which in its extreme
amplified waveform will be altered in a could give rise to a guitar input producing a
similar way as when tone controls are used. 'piano' sound output.

5.2.4 Non linear distortion or clipping (THD)

When the amplifier is overdriven, a clipping

of the output voltage is likely to occur. This
effect, called non-linear distortion,
happens when the input signal exceeds the
dynamic range of the amplifier. When the
voltage is clipped, the normal curve of the
signal wave is squared off, producing extra
harmonics of the fundamental. This is
commonly referred to as Total Harmonic
Distortion (THD).

The result is an audible change, making the

sound uncomfortably raw. Another problem
occurs when the current continues to rise,
causing too much energy to be fed into the
loudspeakers (beyond their Power Handling
Capacity (PHC) limits), which could cause
them to be damaged.

5.2.5 Rated Output Power

Rated Distortion Limited Output Power is the roughly 10 times that of the average power.
power which the amplifier is capable of This is called the 'rated power' of an amplifier.
dissipating in the rated load impedance, at a A 100 W amplifier, for instance, having a input
given frequency or frequency band (1 kHz), sensitivity of 100 mV, will produce 100 W
without exceeding the rated Total Harmonic output when the input voltage reaches 100 mV.
Distortion (THD).
Emotional speech, or certain passages of
music, can cause pronounced audio signal
peaks. Such

instantaneous features of speech and music

have to be reproduced without distortion.
Generally, allowance must be made for speech
attaining voltage peak values of approximately
three times its average. This may be
expressed as 20 Log 3 = 10 dB. 10 dB, as a
power ratio, means that the peak power is

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This 100 W is the maximum output power average output power will only be 10 W
which the amplifier can produce whilst still (allowing up to 100W for peaks). This means
keeping distortion below its specified limit. that, on average, an amplifier normally
Under normal conditions, however, the operates at only one tenth of its rated (or peak)
average input voltage will only be 33 mV value. In our example this will be 10 W.
(allowing up to 100 mV for peaks) and the

5.2.6 Temperature Limited Output Power (TLOP)

The IEC 65 standard states that an amplifier, This means that 100 W amplifiers, located in
running under worse case conditions, should at ambient temperature of 45° C, with + 10%
least be able to run continuously at 12½% of mains over-voltage, stacked on top of each
its Rated Output Power without any other, in a 19 inch rack frame, should be able
components overheating. to run continuously for 24 hours per day at
12.5 W average power without overheating.

5.3 Voice Evacuation Amplifiers

Redundancy

To implement spare amplifiers in a public

address and voice evacuation system is
needed as to make sure that the system will
operate when a main amplifier fails.
Change over relays will see to it that a
spare amplifier will automatically replace
the main power amplifier in case of a failure
is detected in the main power amplifier.

Supervision

Amplifiers used for voice evacuation to-ground and short-circuit detection

purposes have continuous monitoring short- functions.

Backup Power

In case the main power fails in emergency backup power can be based on a power
situations, the systems amplifier must generator, but mostly a battery backup
switch over to a backup power source. The power is used at 24VDC or 48VDC.

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5.4 Amplifier / loudspeaker interface

In order to interface public address and voice loudspeakers changes, it is very difficult
evacuation system loudspeakers with power indeed to match them to the power amplifier.
amplifiers the 100 Volt line matching principle In this type of situation, or in an application
is used. requiring long loudspeaker cable lengths, the
100 Volt line matching system is used.
If the load is always constant, the
loudspeakers can be connected in a In the 100 Volt line matching system,
series/parallel arrangement to exactly match transformers, which are mounted in the
the amplifier's low output impedance. power amplifiers, are tapped to step up the
However if the loudspeakers differ in power output voltage of the amplifiers from a low
and impedance, or if the quantity of voltage to 100 Volts (or 70 or 50 Volts).

Transformers, mounted on the loudspeakers, then reduce this again to the original low voltage,
acceptable to the loudspeakers. This system gives great flexibility in the design and use of public
address systems for the following reasons:

→ By increasing the output power voltage of an amplifier, the amount of current (measured in
amps) involved is reduced significantly. This means that even when high power amplifiers are
used, line losses are kept low, and heavy duty cabling is not required.

→ Due to these low line losses, extremely long cable lengths are possible. This is a very
important factor in a public address installation.

→ All loudspeakers may be simply connected in parallel.

Voltage Tap on the amplifier

Loudspeakers connected to the 100V tap on

the amplifier's line matching transformer are
used at full power, whereas if they are
connected to the 70V tap, only 1/2 of their
rated power is used. This means that the 70V
tap enables the amplifier to power twice as
many loudspeakers, with each loudspeaker
producing 1/2 of its potential power. Similarly,
the 50V tap allows loudspeakers to use 1/4 of
their rated power, so that the amplifier is able
to power 4 times more loudspeakers, with
each producing 1/4 of its potential power.

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Voltage Tap on the loudspeaker

The transformers fitted to loudspeakers have common amplifier, and it is desired to have
similar taps, but in this case the actual power one of them producing less volume than the
which the loudspeaker will draw (e.g. P, P1/2, others, then it is a simple matter of
P1/4, or 6W, 3W, 1,5W), instead of the connecting the signal to either the 1/2 or the
voltage, is printed beside each power (+) tap. 1/4 power (+) tap. This would reduce the
A reduced loudspeaker volume can be set by output of the loudspeaker by 3dB or 6dB
using these taps. For instance if the same respectively.
type of loudspeakers are powered from a

The total power drawn shall not exceed the amplifier’s rated power

When using the 100 Volt line matching simply adding the Rated Power of the
system, the Rated Power of the amplifier connected loudspeakers together, taking into
corresponds to the Rated Load Impedance of account the difference in power drawn when
the loudspeaker network. The total rated using the loudspeaker power taps.
power required should be calculated, by

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6. Cable lengths.

The maximum permissible cable lengths per size of cable are shown in the accompanying graph.

Example:
2
Assuming an amplifier of 100 W tapped at 100 V and using a cable of 2x0.75 mm . The length of the
cable should not exceed 250 m. The values refer to a 10% voltage drop, with the entire load
concentrated at one end of the cable. The lengths can be doubled when the load is distributed
evenly along the cable.

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7. Speakers

The loudspeakers used in the audio reproduction chain are a vital factor in determining the overall
quality and success of a sound system. Because of this, it is vital to understand the different types of
available loudspeakers and their particular strengths and weaknesses.

7.1 Theory of operation

An amplifier is used to boost the low level constantly reverses the magnetic forces
audio signals (such as those from a between the voice coil and the permanent
microphone) to amplitude capable of driving magnet. This pushes the coil back and forth
a speaker. Because the audio signal changes like a piston. When the coil moves, it pushes
polarity from positive to negative, the current and pulls on the speaker cone.
going through the speaker moves one way
and then reverses and flows the other way. This vibrates the air in front of the speaker,
creating sound waves that are in proportion
This alternating current causes the polar to the original audio signal.
orientation of the electromagnet to reverse
itself many times a second corresponding to
the frequency of the input sound. The
electromagnet is positioned in a constant
magnetic field created by a permanent
magnet. The positive end of the
electromagnet is attracted to the negative
pole of the permanent magnetic field, and the
negative pole of the electromagnet is repelled
by the permanent magnet’s negative pole.
When the electromagnet’s polar orientation
switches, so does the direction of repulsion
and attraction. This alternating current

7.2 Loudspeaker types

7.2.1 Standard loudspeaker cabinets

Standard (infinite baffle) loudspeaker volume. Normally, a large sealed enclosure

cabinets, are in principle a sealed box will provide better bass response than a
containing 1 cone loudspeaker, and have a small one.
typically wide dispersion pattern. Their
shape makes them convenient for mounting
on walls or pillars, or suspending vertically
from the ceiling, to give a wide beam of
sound.

The bass response of the sealed enclosure

is very much dependent on its inside

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7.2.2 Ceiling loudspeakers

A ceiling loudspeaker is a cone loudspeaker

mounted on a front panel, which may be
recessed into a ceiling or hollow wall. They
can be spaced at regular intervals to give a
fairly even coverage of sound.

A common used calculation-method leads to

the mutual distance between the speakers:

D = 2 H tan (α/2)

• H = Ceiling
height to Ear
height

• α= opening
angle at 4 kHz

Total number of
the speakers:

n = Area / D2

The table shows the level variations which Further it is difficult to obtain good results
can be expected for different opening from a ceiling loudspeaker system in rooms
angles. with a reverberation time of more than 2
seconds.
Here caution should be taken when
mounting these units in particularly high
ceilings (> 5 meters) and in noisy
environments. The level of sound reaching
the listeners may be unacceptably low, due
to the distance involved, and the limited
maximum SPL available from the units.

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7.2.3 Sound columns

Sound columns are a group of loudspeakers

mounted close together in a vertical array.
Due to an interesting acoustical
phenomenon, though the beam of sound
emitted horizontally is approximately the
same as a normal cone loudspeaker, the
beam of sound emitted vertically is narrow
(10-15°) and therefore very directional,
especially at higher frequencies.

Column loudspeakers are particularly useful

in situations where a great degree of control
is required over the vertical spread of sound,
and no spill of sound is acoustically allowed.
A typical instance would be in reverberant
environments (e.g. churches) where it is
desirable to beam the sound down onto the
listeners, without it reflecting off hard walls
and ceilings.

7.2.4 Horn loudspeakers

Because the diaphragms are normally

mounted in molded plastic or metal folded
horn enclosures, they can be easily
rendered weatherproof, which allows them
to be used outdoors and in dusty and humid
environments. They may be mounted on
masts or higher buildings and/or arrayed in a
column to produce a directional vertical
beam.

The diaphragm loudspeakers used in public

Horn (or 'diaphragm') loudspeakers, are address installations have the limitation of
different to cone loudspeakers in that the having a fairly restricted frequency range,
sound produced is generated by a small, giving a diminished output at low
thin metal diaphragm, and amplified by the frequencies due to the diameter of the horn
shape and size of a folded horn. They and at high frequencies due to folding of the
produce a very powerful, concentrated, horn. This makes them generally unsuitable
beam of sound enabling them to reach for satisfactory music reproduction, but can
listeners at a great distance. to some degree be compensated for by
combining them with cone loudspeakers.

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7.2.5 Full range high power loudspeakers

Loudspeakers with diaphragms mounted

directly onto the mouth of an exponential
horn are often used as the treble component
of a "full range" multiple loudspeaker
enclosure. The audio signal is fed through a
suitable crossover filter which eliminates the
bass content, which could damage the
diaphragm. These enclosures, often
grouped together in a cluster, are used in
installations to produce full range high power
sound.

7.2.6 EVAC Loudspeakers

Any short circuit or interruption in the voice evacuation message not reaching the
alarm transmission path between the Central audience.
equipment and the loudspeakers shall not
result in total loss of coverage in the
loudspeaker zone served. In other words
failure of a loudspeaker may not result in the

Redundancy

A straightforward solution to implement different loudspeaker lines. One

redundancy on the speaker part of the loudspeaker is fed from the ‘A-system’ with a
installation is the use an A/B setup. Per ‘B-system’ speaker in between each two ‘A-
area/zone two amplifier channels feed two system’ loudspeakers.

Short-or open-circuits

A ceramic terminal block is used to prevent

short-or open-circuits. High temperature
wiring and a thermal fuse (resistor against
150 degrees Celsius) are installed to
disconnect the loudspeaker automatically.

Supervision

Even where faults do not affect the a useful feature used mostly in high end
operation of loudspeakers, any system.
malfunctioning of a speaker or speaker line
must be notified. The supervision of Supervision of the loudspeaker lines can be
loudspeaker lines is obligatory, whereas the done in different ways each with their
supervision of each separate loudspeaker is specific advantages and disadvantages.

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Working Strengths Weakness

• Imprecise because of
Periodically the Back cable capacitance (Fire
Ground Music (BGM) is proof E30 cable has high
Impedance switched off, volume capacitance)
measure- bridge is activated and • Low cost • Imprecise because of
ment an impedance • No extra installation temperature increase
measurement is done • Free topology of during operation
(a.o. Plena with a high (18-20kHz) loudspeaker wiring • Back ground Music
VAS) or low (1-2kHz) interruption
frequency signal. • Sensitive for back ground
noise (loudspeakers
generate voltage)
• Precise
A 20kHz pilot tone is
• measurement of
Pilot tone sent continuously to a
• Total line • Local volume regulators
detection pilot tone detection
• No back ground are not allowed
Continuous board at the end of a
Music interruption • Loudspeaker line
(a.o. Plena loudspeaker line. 2
• Continuous branches not allowed
VAS) wires return a status
information. supervision

Periodically a 20kHz
pilot tone is sent to a • Precise • Only periodically checked
Pilot tone pilot tone detection measurement of • Local volume regulators
detection board at the end of a Total line are not allowed
periodically loudspeaker line. 2 • No back ground • Loudspeaker line
wires return a status Music interruption branches not allowed
information.
• Precise
A 20kHz pilot tone is
measurement of
Pilot tone sent to a pilot tone
Total line • Local volume regulators
detection detection board at the
• No back ground are not allowed
Wireless I end of the loudspeaker
Music interruption • Loudspeaker line
(Bosch line. A 70kHz tone is
• Continuous branches not allowed
Praesideo) returned over the same
supervision
loudspeaker line.
• No extra wiring
• Precise
• measurement of
A 20kHz pilot tone is
• Total line
sent to a pilot tone
• No back ground
detection board at a
Pilot tone Music interruption
speaker, at the end of
detection • Continuous • Local volume regulators
the loudspeaker line or
Wireless II supervision are not allowed
loudspeaker line
(Bosch • No extra wiring
branch. A 70kHz tone
Praesideo) • Supervision
with board address is
returned over the same possible per
loudspeaker line speaker
• Free topology of
loudspeaker wiring

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Principles of Public Address and Voice Evacuation
• Extra installation cost
(loudspeaker has to be
• Low cost
disassembled to mount a
• No back ground
Periodically a Direct capacitor before each
Music interruption
Direct Current is sent through loud-speaker
• No extra installation
current (DC) the loudspeaker line transformer)
• Free topology of
measure- and impedance of all • Loudspeaker
loudspeaker wiring
ment line mounted resistors is measurement not viable
measured • Cable capacitance
for loudspeaker that
is not a problem
cannot be disassembled.
(DC)
• Audibility of low
frequencies decreases

7.2.7 IP rating

A two-digit number (as specified in • The first digit represents protection against
Australian Standards AS 1939 and EN ingress of solid objects
60529) is used to provide an IP-rating to a
piece of electronic equipment or to an • The second digit represents protection
enclosure for electronic equipment. The two against ingress of liquids
digits represent three different forms of
The larger the value of each digit the greater
environmental influence:
the protection from that influence.

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7.2.8 Technical Principles

Stacking of loudspeakers

Each time the input power of a loudspeaker a sensitivity of 99 dB (1W/1m), 2W would

is doubled, the SPL rises by 3 dB. Therefore raise the SPL by 3 dB, to 102 dB; 4W would
if we know the sensitivity of a loudspeaker, it increase it to 105 dB; etc., until it reaches
is a simple matter to calculate its SPL at any maximum rated power.
given power input. E.g.: If a loudspeaker has

If 2 loudspeakers are placed side by

side and given the same input signal
(so that both are in phase, operating
as one unit) the SPL at the listeners
would be 6 dB more than the SPL of
a single speaker. Each time the
quantity of loudspeakers is doubled,
the SPL is increased by 6dB.

If those same loudspeakers are

placed some distance away from
each other (even so small a distance
as 1 meter), there will always be a
shift in phase at the ears of the
majority of the listeners. This causes
the total SPL to increase by only 3
dB, instead of 6dB, each time the
quantity of loudspeakers is doubled.

Sound pressure level and distances

As we move further away from the sound meter; at 2 meters distance the SPL would
source the SPL drops. Again a simple rule is be 106 dB; at 4 meters 100 dB etc. (this rule
in force; Each time the distance from the only deals with direct sound, not taking into
loudspeaker is doubled, the SPL drops by 6 consideration any (indirect)sound returned
dB. For instance, assuming that we have a from reflective surfaces).
loudspeaker cabinet producing 112 dB at 1

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SPL and opening angle

All of these examples so far have dealt with

loudspeakers producing a fixed frequency
tone being measured in line with the
loudspeaker's axis. By looking at the polar
diagram, we can see that the SPL differs
depending on the frequency being
transmitted, and at what angle the listener is
relative to the axis (0o = directly below the
center of the speaker).

The number of degrees between the points where plot lines cross the -6 dB determines the
opening angle. For intelligibility & clarity reasons the opening angle up to 4 kHz is vital.

Resonant frequency

At the resonant frequency the impedance is Damage can occur to the loudspeaker if
very high in relation to the average power is sustained at the resonant
impedance. This varies from cone frequency. So where continuous alarm
loudspeakers (20 Hz to 300 Hz) to horn signals are required, care should be taken to
drivers (200 Hz to 1 kHz). The 'nominal ensure that the frequency of the signal is
impedance' is the impedance of the lowest well above the resonant frequency of the
part of the curve above the resonant loudspeakers used.
frequency (fo) - usually around 400 Hz.

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Sensitivity

The sensitivity level of a loudspeaker is the Because the efficiency of loudspeakers,

loudness expressed in dB (SPL) at 1 kHz horn drivers, columns, etc., vary so much, it
and at a distance, on axis, of 1 m with an is impossible to define the number of
input of 1 W. The importance of this figure loudspeakers required for a room (and the
may be illustrated by examining the effect of amplifiers required to drive them) without
varying the two main parameters, namely, first calculating.
distance and power.

Example

• Assume that it is required to produce a SPL of 80 dB at a distance of 32 m.

• To calculate the required power for the loudspeaker (For simplicity an outdoor situation is
chosen): Reduction in acoustic level due to distance = 20 Log 32 = 30 dB

• To compensate for this reduction, 80 + 30 = 110 dB (SPL) is required at 1 m distance from the
loudspeaker.

• If the loudspeaker has a sensitivity of 100 dB (SPL) the missing 10 dB should be compensated
for with an extra 10W power applied to the loudspeaker.

Efficiency

A loudspeaker's ability to convert electrical percentage figure (values between 0.5-

energy into acoustical energy is defined as 10%). This value is required for calculations
its efficiency, and can be stated as a of the reverberant sound field.

Directivity (Q)

The directivity factor (Q) of a loudspeaker is sound pressure level at the same distance,
the ratio of the mean squared sound averaged over all directions. Q is therefore a
pressure level at a fixed distance, measured measure of the response of the loudspeaker
on axis (which is normally the direction of in a three dimensional plane.
maximum response), to the mean squared

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Principles of Public Address and Voice Evacuation

At low frequencies the radiation of a seven octave bands, this requires the
loudspeaker has a spherical form which processing of more than 2000 readings.
becomes more directional as the frequency Because of this only a few of the leading
increases. This indicates that Q is frequency manufacturers actually quote figures for the
dependent. Since readings are normally directivity factor.
taken in 10° intervals in a sphere, for each of

Standard format
for loudspeaker
technical
specifications,
showing average
performance for
each of the
seven octave
bands.

These specifications are measured in an anechoic room following the procedures defined below:

1. The frequency response is measured The effective frequency range is defined as

on axis (0o) at 5 meters and calculated being the range between those points at which
to 1 meter: the level drops by 10 dB.

• with “slow” damping using a gliding

tone and/or a 1/3 octave wobble
tone.

• in 7 octaves, using a stepped pink

noise measurement signal.

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2. For enclosures with single loudspeakers, enclosures with asymmetrical or multiple

polar diagrams are measured using pink loudspeakers the directivity balloon is
noise. This is done in octave steps with measured using a pan and tilt device.
centers at 125 Hz, 250 Hz, 500 Hz, 1000 The definition is every 100 for all 7
Hz, 2000 Hz, 4000 Hz, and 8000 Hz. For octave bands.

3. Using the measurements in 1. and 2., 4. Using the measurements in 2., the
the software package EASE is used to horizontal and vertical opening angles
calculate the “Q” and “Efficiency” (-6 dB) are determined for all relevant
values for all relevant octave bands. octave bands.

5. The Power Handling Capacity is

determined by applying the IEC pink
noise test signal shown below for 100
hours. After this test the loudspeaker
should still be able to perform
according its specification.

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8 Equalization and Acoustic Feedback

The increase in sound level which a sound The use of sound equalization to reduce
amplification system can give to a acoustic feedback contributes toward the
performance in an auditorium is limited by comfort of both performers and audience and
acoustic feedback. In many cases, e.g. when will enhance the acoustic quality and increase
the performance requires a long microphone the overall system gain.
distance or with a somewhat noisy audience,
the feedback limit prevents an adequate Room conditions can also reduce intelligibility
sound level being produced for comfortable as they “color” the sound by changing the
listening. This situation can be annoying to frequency response. This effect can also be
both performers and audience because of an corrected by equalization.
insufficient sound level or when the system
amplification is increased, spurious ringing
sounds.

The acoustic feedback loop

The total system loop contains basically a sequence of the following elements:

• microphone
• amplifiers with volume/tone control
• loudspeakers
• acoustic transmission link between loudspeaker(s) and microphone

The acoustic transmission link consists of one

(or two) direct path between the
loudspeaker(s) and the Microphone which is
maintained by what is called the “direct sound
field”, and many other paths caused by
reflections and multi-reflections which are
maintained by what is called the “diffuse
sound field”.

Resonant acoustic feedback

Acoustic feedback is spontaneous oscillation a) the phase angle of the transmission

caused by the transmission of sound radiated through the acoustic feedback loop
by the loudspeaker(system output) back to the equals zero, and
microphone (system spontaneous oscillations
input). When the gain of the sound system is b) the sound from the loudspeaker re-
gradually increased, a point will be reached enters the microphone louder than the
where spontaneous oscillations (howling) start original sound (loop gain ≥1)
to occur or resonant acoustic feedback can
occur at any frequency for which:

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This cycle repeats itself, with increased howling sounds. To avoid ringing sounds
amplification until the sound reaches the during speech or musical performances, the
system’s maximum loudness or until someone gain has to be reduced to approximately 6 dB
turns down the volume! Even though a sound below the level at which spontaneous
system is adjusted just below its critical gain, oscillation begins, this is called Feedback
feedback will prolong the signal components Stability Margin (FSM).
at this critical frequency, producing ringing or

Principles of equalization

The ideal in any audio system is to obtain a acoustic link, changes are introduced to the
flat frequency response over the complete feedback frequency response by the very
audio frequency band. When considering the nature of the auditorium. The cancellation of
sound system equipment alone, a flat these changes in the frequency response,
frequency response can be achieved within whether they be peaks or dips, is called
very fine limits, but when taking the sound “Equalization”.
system as a whole, with its associated

Using a measuring set-up as

explained on the next page we can
obtain e.g. the following loop
response:

The dominant frequency where the

acoustic feedback is likely to occur is
160 Hz and secondly 3.4 kHz.

By equalizing the loop response now with a automatically by a so called intelligent

“mirror imaged” filter response, the overall feedback exterminator which work with a
gain can be increased. In order to maintain number of narrow band filters adjusted
the optimum signal to noise ratio, increasing dynamically at the critical frequencies and
the gain at dips in the frequency response, as maintaining a FSM of 6dB.
well as reducing the resonant peaks, should
be considered. This can be done manually by
means of a graphic or parametric equalizer or

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It is impossible to lay down hard and fast rules possible gain for all frequencies and preserve
as to which equalization method should be the signal to noise ratio. A listening test after
used, as the requirements will be vary from equalization is important because a flat loop
one auditorium to another. The prime response is not always a flat listening result, a
objective is to obtain a flat frequency high frequency roll off is sometimes required
response of the loop to obtain maximum (3 dB/octave > 1 kHz).

Loudspeaker equalization

In a Public Address system where back The most convenient method is to inject pink
ground music is played, measurements noise in the sound system’s line input and
concentrate on the loudspeaker reproduction. measure with a 1/3 octave Real Time
In this case a calibrated measuring Analyzer on the audience position. A 1/3
microphone at the audience position is used, octave graphic equalizer is the easiest to
and equalization is done for the power adjust but difficult to hide for unauthorized
amplifier, loudspeaker(s), and room. tempering.

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9 Time Delay

When a public address system is set up in a To overcome this disturbing effect, the sound
large stadium bowl, railway station or tunnel, from each (group of) loudspeaker(s) must be
where the aural announcement origin is delayed using time delay equipment (in high
located at the closest loudspeaker, but is then end equipment this is incorporated in central
followed by arrival of sound from the other control and amplifying equipment).
loudspeakers, causing echoes and
reverberation.

To overcome the disturbing effect, the sound and will benefit the intelligibility considerably.
from each (group of) loudspeaker(s) must be An effective way of doing this can be to use the
delayed using time delay equipment. If the loudspeakers located in the middle of the
timing is set properly, the sound will be platform as the starting point.
synchronized with the furthest loudspeakers

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10 Automatic Volume Control

Automatic Volume Control (AVC) regulates the maximum listening level. If the loudspeaker
loudness of a P.A. announcement relative to system is set up so that a maximum of 89 dB
the ambient noise level. This guarantees SPL can be achieved, then a control range of 9
maximum intelligibility and minimum dB would be the right choice.
annoyance.
An AVC unit with 21 dB control range would
The ambient noise level is continuously only be used in PA systems which can produce
measured by a microphone connected to the a maximum level of 101 dB(SPL), being 21 dB
sensor input of the AVC unit, which uses this above the comfortable listening level of 80
measurement to set the attenuation of the dB(SPL).
signal path. During periods of low ambient
noise, the Public Address & Voice Evacuation The AVC unit is factory pre-set, therefore only
system’s gain is reduced by the AVC-unit, and the sensing input microphone gain in the
during periods of high ambient noise, the corresponding
Public Address & Voice Evacuation system’s
Loudspeaker zone and the reset time
gain is restored to its nominal maximum. A
(blocking) needs to be adjusted. If a
blocking circuit ‘freezes’ the input sensor while
microphone is located inside the loudspeaker
an announcement is being made, ensuring that
zone to which it is addressed, the gain should
the announcement itself is not measured by
be carefully set to avoid acoustic feedback.
the unit as ambient noise.
The system should be checked during periods
The control range of the AVC, with attenuation of high ambient noise and low level talking into
values from 6 to 21 dB, depends on the the microphone in order to ensure that no
maximum loudness of the sound system. 80 acoustic feedback, automatic attenuation
dB (SPL) is regarded as a comfortable (AVC), or limiting (Call station) occurs.

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11 The Acoustic Environment

The characteristics of sound and the way it is a large reverberant church or to a heavily
transmitted are very much altered by the damped lecture room.
environment in which it is generated. The
same audio signal would sound quite In general, it is possible to differentiate
different in a sports stadium as compared to between two situations: the outdoor and the
indoor environment.

In both situations though we are striving primarily at:

1. Speech Intelligibility - delivering the 2. Quality of Reproduction - delivering e.g.

message to the ears of the listener music to the ears of the listener as
clearly. unchanged as possible.

11.1 Outdoors

In the outdoor environment several factors must be considered which influence sound
reproduction and reception:

• Power
• Directivity
• Distance
• Reflection
• Absorption
• Air absorption
• Echoes

Power

Each time the input power of a Using our reference

loudspeaker is doubled, the SPL rises of 100 dB(SPL), for
by 3 dB. The effect, at a distance of 1m, an increase of 12 W
is shown in the table, which lists the the calculation is:
increase in SPL with doubling of power,
from a nominal value of 100 dB (SPL) 100 + 10 Log 12 =
100 + 10.8 = 110.8
Intermediate powers may be accounted dB (SPL)
for by: dB (SPL) at measured power =
SPL 1.1 + 10 Log P/P0 where:

• SPL 1.1 = sensitivity of Each time the input power of a loudspeaker

loudspeaker in dB (SPL) for 1 is doubled, the SPL rises by 3 dB. The
watt at 1 meter effect, at a distance of 1m, is shown in the
• P = power (W) table, which lists the increase in SPL with
doubling of power, from a nominal value of
• P0 = reference power (1W)
100 dB (SPL)

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This SPL increase

of 10.8 dB for a
power increase of
12 W can also be
seen in the
accompanying
table.

Directivity

A fundamental difference between opening angle or to concentrate a beam of

loudspeaker types is their 'opening angle'. sound in a particular direction.
This is the dispersion (measured as an
angle) of sound which radiates from the front This latter would be important where an
of the speaker. unnecessarily wide spread of sound is not
only wasteful in amplifier energy, but could
Depending of the environment and the reflect off nearby buildings, or disturb people
particular application needs, it may be in neighboring areas.
necessary to use loudspeakers with a wide

Distance

When sound is reproduced in an outdoor 6 dB(SPL) each time the distance is

situation, without any objects to cause doubled. The table below shows SPL
reflection, the listener hears only direct decrease with the doubling of distance, from
radiation. The sound pressure level drops by a nominal value of 100 dB(SPL)

Example:

Assume that a loudspeaker source has a

sensitivity (SPL1.1) of 100 dB(SPL). An
input of 1 W gives the following results:

For intermediate distances:

dB(SPL) at measured distance = SPL1.1 - 20 Log r/r0

where:

• SPL1.1 = sensitivity of Using the nominal value of 100 dB, the

loudspeaker in dB(SPL) 1W;1m calculation of the SPL at 25 meters is:
• r = measured distance (m)
• r0 = reference distance (1m) 100 - 20 log 25 = 100 - 28 = 72 dB(SPL)

The SPL decrease of

28 dB at a distance
of 25 meters can
also be seen in the
accompanying table.

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Variations of both distance and power

Assume that a loudspeaker has a Generally we can calculate as follows:

sensitivity of 100 dB. To calculate the
dB(SPL) at 26 m with an input of 10W: • Ldir = Ls + 10 Log(Pel) - LQ - 20
Log(r) Ls = SPL1.1 = SPL value for
• At 26 m the loss in dB(SPL) = 20 1W at 1m on axis.
log 26 = 28.3 dB • Pel = power consumption of
• And at 10 W, gain in dB(SPL) = 10 loudspeakers (W)
log 10 = 10 dB • LQ = on/off axis level difference
• r = distance from the source
The total effect of both variations is simply
their algebraic addition:

• 100 - 28.3 + 10 = 81.7 dB(SPL)

Reflection

Although the effect of reflection is mainly of original. Knowing the speed of sound in air
concern in an indoor situation, reflections to be 340 m/s, then the time difference of 50
from buildings outdoors give distinct and ms is equivalent to a distance of 17 m. If the
very disturbing echoes. If the time delay difference between the direct and the
between the original sound and the reflected indirect distances is significantly shorter than
sound is more than 50 ms, the listener will 17 m, the reflected sound will have the effect
be able to hear, and to recognize, a of reinforcing the direct sound, rather than
reflected sound as a whole "echo" of the causing an echo.

Ambient Noise

The perceived quality from a sound When the sound level of a source is being
reinforcement and/or public address measured in a situation where ambient noise
distribution system can be particularly is present, it is necessary to subtract the
affected by ambient noise. The constant ambient noise level reading from the
sound of passing traffic, the rumble of heavy combined (total) reading in order to find the
industry or even the hum of conversation actual level of the source alone. If this is not
from a large crowd can create a significant done it is not possible to measure the source
ambient noise level, which must be level accurately.
compensated for.

This is calculated by:

• Ls = 10 Log [10 L1/10 -10 L2/10]

• L1 is the reading taken of the source and the noise combined (e.g. 60 dB(SPL)) and,
• L2 is the reading of the noise alone, with the source shut off (e.g. 55 dB(SPL)).

In this example the level of the source is: Ls = 10 log [106 - 105,5] = 58,3 dB (SPL)

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11.2 Indoors

When designing a sound system for indoors, This means that, in a reverberant
the situation is made difficult by a number of environment, with increasing distance, we
problems which must be taken into encounter two problems at the listeners:
consideration.
• A decreasing original (direct) speech
Because the listener is often positioned at spectrum (SPLdir)
some distance from the source of the sound,
high frequency signals are absorbed by the • A reverberant low toned
air, while the lower signals activate indirect/reflected speech spectrum
reverberation as they bounce off hard walls (SPLrev). This means that the listeners
and ceilings. may hear everything loudly, but the
consonants in the speech are hidden or
masked by the reverberation, causing
low speech intelligibility, so that they
cannot understand what is being said.

11.2.1 Direct Field and reverberant Field

Reflection & Absorption

When a sound source is in a room and Most of the building materials have
enclosed e.g. by walls and a ceiling, these measured absorption coefficients ( inc) and
surfaces will partly reflect and partly absorb reflection coefficients (r).
the sound. The intensity of the reflected
sound wave ( ref) is smaller than the • α + r = 1.
incident one ( inc), a fraction of the incident
If all the sound is reflected (r = 1), no sound
energy is lost during reflection, or:
is absorbed by the material (α = 0).
• ref = (1 - α) inc

• α is called absorption coefficient

Reverberation

If sound is generated in a room, part will these repeated reflections is called

travel directly to the listener; more will arrive reverberation, which leads to the build-up of
after having been reflected and still more diffuse sound throughout the room, called
after successive reflections. The effect of the reverberant field.

The actual level of the reverberant field is

determined by three factors:

• the nature of the sound source

• the physical volume of the room
• the reverberation time.

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Reverberation time

The reverberation time (T) of a room is a • the reverberation time in a room is

measure of the time taken for the sound the same whatever the position of
level of the reverberant field to fall by 60 the sound source;
dB. The following points regarding
reverberation time are assumed: • the reverberation time in a room is
the same wherever the listener
happens to be;

• the lack of intelligibility in a room is

almost always due to a long
reverberation time;

• reverberation time is determined by

the room volume, and total amount
of sound absorption in it.

The reverberation time according to Sabine:

The effects of the number of people in the having the same absorption as a person
room should normally be taken into actually in the seat. In other situations like
consideration. In many theatres and airports where the reverberation time of the
cinemas however, the effect of the variation empty hall is known, the influence of the
in audience numbers is minimized by the audience can be calculated by:
use of plush sound absorbing seating,

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Calculation example for determining the Reverberation Time (T)

Calculation of Direct and Indirect Sound Fields

It is important to have a good

understanding of the different
sound fields in a room. Early
sound carries the intelligibility, late
sound gives the disturbance. Early
sound is experienced by our ears
as the sum of all speech related
sounds arriving in a time window of
20-30 ms. This is the direct sound
coming straight from the source(s)
plus the indirect sound due to
reflections as long as they are
within the time window (split time).

The level of this early useful sound (Ldir) can be calculated like to following:

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Calculation of Reverberant

All the speech related sound which arrives (Lrev) depends of the source(s) , the
later than 20-30ms is regarded as useless volume and the reverberation time of the
and disturbing and consists of a chaos of room. The following formulas can be used
reflections and is called reverberation. The to calculate the reverberant sound field:
level of this reverberant disturbing sound

11.3 Speech intelligibility graph STI & RASTI

This diagram helps to make a direct

translation from the level difference between
useful Direct sound and disturbing
Reverberant sound into Speech Intelligibility.
The useful sound level will vary at different
positions in the room, depending on
distance, angle and useful reflections.

The corresponding formulas are used in the

example below, which is a sound
reinforcement application with two
loudspeakers (1) & (2).

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Example

After calculating on a particular position in time (e.g. T = 3s) and read the Speech
the room this level difference we enter the Transmission Index (STI) value at the right
chart at the bottom and go up to the edge of the chart. (Example: 4dB > STI =
intersection with the actual reverberation 0.585 )

11.3.1 Articulation Losses of consonants in speech. (% ALcons)

Since most of the information in a language

is conveyed by the consonants, intelligibility
may be expressed in the percentage
articulation loss of consonants (%Alcons).

Source: Audio Engineering Society Journal

The %ALcons is the difference between 1. The reverberant field "Lrev" depends on
the direct and reflected field levels, the total acoustic power of the source(s);
calculated as a function of the the volume of the room; and the
reverberation time. If these figures are reverberation time.
known, the accompanying graph makes
it possible to quickly calculate %Alcons. 2. The direct field "Ldir" depends on the
Assuming that a loudspeaker is radiating sound level at the source(s) at 1m
sound in a room with volume V and pointing at the listener, and on the
reverberation time T. distance from the source(s).

3. The difference between the calculated

Ldir and Lrev , expressed in dB (SPL), is
a reliable measure for the expected
speech intelligibility.

4. Using this difference, we enter the chart at

the bottom and go up to the intersection
with the corresponding Reverberation
Time line.

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If we then go right we will find the 1. The ambient noise level is measured
%ALcons figure at the right edge of the using a sound level meter with A
chart. If the difference between the speech weighting and set to Fast reading.
signal peak level and the ambient noise
level is smaller than 35 dB, and no limiter 2. The signal level is determined by: a)
is used, the speech intelligibility will be calculating, using the sum of Ldir and
effected. Lrev. b) measuring in dBA (Fast)with 10
dB added. c) measuring directly in dBA
(peak or peak hold).

3. Using the %ALcons value, determined

from the previous chart, enter at the left
side and follow the sloping line down,
until it intersects with the vertical S/N
ratio line. From this point go right to
read the new %ALcons value at the
right edge of the chart.

11.3.2 Speech Transmission Index (STI & RASTI)

A method of calculating and measuring sound system is used for Public Address
speech intelligibility has been developed, and Voice Evacuation a minimum STI value
called the Speech Transmission Index (STI) of 0.5 is obligatory.
method, which evaluates the intelligibility
over 7 octave bands from 125 - 8000 Hz. It Portable (battery operated) measuring
basically stands for the speech signal equipment, manufactured by Bruel & Kjaer,
transmission between the signal source uses the so called Rapid Speech
position and listener position. The STI Transmission Index (RASTI) method of
values are theoretically between 1(ideal) calculation, and restricts the measurements
and 0 (bad) in practice between 0.75(ideal) of the speech transmission to the 500 Hz
and 0.25(bad). In installations where a and 2 kHz octave bands only, instead of all
7 octaves.

11.3.3 Subjective %ALcons and RASTI requirements.

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11.3.4 Converting RASTI to %ALcons

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12. Loudspeaker placement and coverage

A few practical considerations must be taken into account when selecting, placing and aiming a
loudspeaker in a sound system design.

→ The loudspeakers must be positioned in → In installations with multiple

such a way that they are able to produce loudspeakers, spacing the loudspeakers
an even spread of sound, reaching all less than 15 meters apart will help
audience areas of the room with minimize echo otherwise proper delayed
adequate loudness and clarity. If this is signals should be applied.
not so, some listeners could be exposed
to an uncomfortably high SPL, while → For ceiling systems the spacing of the
others may have difficulty in actually loudspeakers should be determined by
hearing the audio signal sufficiently. looking at the covered areas (-6dB) at 4
kHz. The audience area divided by this
→ Speech requires generally a good coverage area gives the number of
transmission and reproduction of the speakers. It means that the audience will
500 Hz to 5 kHz frequency band, while hear the announcements at about the
music requires at least 100 Hz to 10 kHz same level for the required spectrum.
to give satisfactorily results. This should This can be done by using the software
be taken into consideration in selecting a package “Ease Address”.
loudspeaker type.
→ Background noise, or ambient noise,
→ For speech applications, up to the 4 kHz can make a great deal of difference to
octave band is essential for the the level required for an adequate
annunciation of consonants, and intelligibility, especially in noisy
therefore intelligibility. Therefore we use environments such as factories or
the loudspeaker opening angle data at 4 airports. To keep the level more than 15
kHz for the calculations for equal dB louder than the ambient noise, the
coverage. use of proper call stations with build-in
compressor/limiter is required.

Given the specifications of the loudspeakers ceiling, are calculated for the important
used it is possible to predict the performance frequencies (125 - 250 - 500 - 1000 - 2000 -
of a sound system before the system is 4000 - 8000 Hz) at any point in a room or
installed / purchased. By using the software area.
package "EASE" the level of the direct sound
(SPL) as received from the loudspeakers and Also the intelligibility is calculated to verify
early reflections from side walls and/or that the listeners can hear the reproduced
sound clearly.

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12.1 Loudspeaker design steps

Design steps:

1. Select the correct loudspeaker type(s).

2. Select the optimum loudspeaker position(s).

3. Select the best aiming points.

4. Check the coverage at 4000 Hz.

5. Calculate the SPLdir on the aiming point(s).

6. Calculate the SPLdir on the - 6dB points.

7. Select the Powertapping(s).

8. In reverberant rooms calculate SPLrev.

9. Check the intelligibility in STI or Alcons(%). For a voice evacuation system STI > 0.5

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