PA System Components

• PA Systems
• Loudspeakers
• Speaker Enclosures
• Crossovers
• Amplifiers

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PA Systems
The objective of a PA system is to amplify sound and deliver it effectively to the ears of the
audience. It may seem that this would be fairly easy to do, but can be extremely difficult to
accomplish.

What should a good PA system be?

• Clean – free from distortion
• Clear – free from noise
• Intelligible – recognisable and comprehensible
• Natural – we want it to have all available frequencies – to replicate the sound as
accurately as possible
• Balance – all separate elements should contribute
• Evenly dispersed – the whole audience should hear it with all the above elements at
an adequate sound level

System Concept
The simplest of PA systems would have 2 transducers and an amplifier (see diagram
below). A transducer is any device, which changes one form of energy into another. For
example a microphone will change sound waves – acoustical energy – into an equivalent
audio signal – electrical energy. The loudspeaker of a system would then change this
electrical energy back into acoustical energy so we can hear it.

An amplifier makes an electrical copy of an electrical signal, and increases the signal
strength so it can supply the loudspeakers. This gain increase can be as much as
1,000,000,000 or more times the strength of the original input signal.

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Fig 1. Simple PA System

In reality, we would have more components to our system to manipulate the sound – this
is called signal processing:

Fig 2. More complex PA System

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Input transducers (i.e. microphones), mixers and outboard gear will be covered in separate
modules in details, including micing and mixing techniques. In this module we will
concentrate on the output stage of the PA system – the loudspeakers and amplifiers.

Loudspeakers
A loudspeaker is a device, which converts electrical signal into sound – acoustical energy.
Loudspeakers are the most variable, limiting element on the faithfulness of reproduced
sound, because they are mechanical. The other stages in sound reproduction are mostly
electronic, and these electronic components are highly developed, so are more accurate,
keeping the output signal as close as possible to the original input signal. The
loudspeaker involves electromechanical processes where the amplified audio signal must
move a cone, or some other mechanical device, to produce the sound like the original
sound wave. This process has many complications and difficulties, so it is important that
you understand the basics of loudspeakers, loudspeaker types and enclosures.

Types of Loudspeaker
There are 3 main types of loudspeaker:
• Dynamic Loudspeaker
• Electrostatic Loudspeaker
• Magnetic Ribbon Loudspeaker

Dynamic Loudspeaker
A speaker uses the principle of magnetism. A magnet has 2 opposite forces contained in
itself, generally called north and south poles (or positive and negative). Unlike poles
attract, while like poles oppose each other.

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Fig 3. Electromagnetism

Speakers use a fixed, permanent magnet and a movable electromagnet, which is created
by current flowing through a coil of wire. The current travelling through coil is AC
(alternating current) and it changes direction according to the frequency of the signal (the
sound being amplified). As the current passes through the coil in one direction, it creates
a positive magnetic field. The two magnetic fields (the fixed magnet, and the created
electromagnet) repel each other, so the coil begins to move out, away from the magnet.
When the AC signal changes direction the coil is forced to move in the opposite direction.

The coil is attached to a paper cone and as the coil moves in and out, as does the cone,
creating the particles in the air to move, therefore creating sound waves.

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Fig 4. Dynamic Loudspeaker components

Components of Dynamic Loudspeakers

Magnet
The magnet size determines how strong a magnetic field can be generated. The larger
the magnet, the more efficient the speaker is because of more control over the voice coil.
It is however heavier and more expensive.
Voice coil
The voice coil is wire windings around a cylinder (called a voice coil former or bobbin),
often made of plastic, aluminium, paper or fibreglass. The coil is continually handling large
amounts of current from the amplifier, so it is important that high power-handling speakers
must incorporate materials in the voice coil to add heat sinking and/or mechanical integrity.
It must also have sufficient strength and temperature resistance to maintain the speaker’s
roundness, so it reproduces the sound accurately. The wire can be made of aluminium or
copper, and it is either round wound or flat wound. Aluminium is lightweight, so it moves
easily, but copper is a better conductor of electricity, therefore having less power loss.
Gap
The gap is the space in the magnet assembly for the voice coil to move in. By narrowing
the gap, efficiency is increased. However the suspension has to be very accurate so that
the voice coil doesn’t touch anything, or the speaker will short out.
Pole Piece and Plates
The pole piece is sandwiched between the front and rear plates, which are made of metal.
The plates become magnetised and direct the magnetic field so it is concentrated into the

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gap, where the voice coil is moving. It also transfers heat into the magnet from the voice
coil to avoid overheating of the voice coil.
Cone
The cone is made of paper or synthetic material, and is glued to the voice coil. The cone
is used to push the air, to recreate the sound. The shape of the curve of the cone can
affect the frequency response, and will also affect the rigidity of the cone.
Surround
The surround connects the cone to the speaker basket helping the speaker maintain its
round shape, as well as determining how far out the speaker can travel.
Spider
The spider connects the voice coil or the beginning of the cone to the basket to centre the
voice coil in the gap. It has rolls or pleats to allow the speaker to move, whilst preventing
excessive excursions.
Basket
The basket is the framework of the speaker’s components.
Dust Cap
The dust cap protects the voice coil and the gap from dirt. It can be made from aluminium,
cloth or paper.
Vent
Many speakers have an opening in the back of the magnet assembly called the vent to
allow additional cooling of the gap area.

Electrostatic Loudspeaker
Electrostatic loudspeakers operate by interacting with charges of static electricity to create
the movement necessary to produce sound waves.

An electrostatic loudspeaker has 3 conductive plates, 2 outer plate which are fixed, and a
middle plate which is suspended so it is free to move. The middle plate is the diaphragm of
the loudspeaker, like the cone of the dynamic loudspeaker, usually consisting of a plastic
sheet impregnated in a conductive material such as graphite. If the diaphragm is very
positive and the outer plates are slightly positive and slightly negative relative to each
other, the middle plate will be attracted to (and be pulled towards) the more negative plate

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while it is repelled (and therefore pushed away from) the more positive plate. This system
is known as push-pull.

Magnetic Ribbon Loudspeaker

As we know from our dynamic loudspeaker, if you put current through a piece of wire, you
generate a magnetic field around it. If that wire is suspended in another magnetic field,
then the field that you generate will cause the wire to move. Ribbon loudspeakers use the
same principle. A piece of corrugated aluminium is suspended in a magnetic field and a
lead wire is connected to each end of the ribbon. When we apply a current through the
wire and ribbon, we generate a magnetic field and the ribbon moves. The ribbon is
effectively the diaphragm of the loudspeaker, moving back and forth with AC current.

Advantages and Disadvantages of loudspeaker types

Loudspeaker Advantages Disadvantages

Dynamic • Rugged • Needs lots of energy for low
• Can be loud frequencies, so have to be larger
• Easy and Cheap and heavier.
to manufacturer • Magnets are heavy.

Electrostatic • Diaphragm is • Need a very big polarizing

very thin – good voltage on the diaphragm
at high (5000V).
frequencies • Needs a very small gap between
• Good transient the plates, but if it is too small it
response can’t have a big enough
excursion (movement) of the
diaphragm for low frequencies.
• Expensive!
Magnetic Ribbon • Low mass – • The ribbon cannot be large or it’ll
good high break, therefore it is not good for
frequencies low frequencies or high Sound

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 Pressure Levels (SPLs).
• Magnets produce a very strong
magnetic field making them very
heavy
• The driver impedance is low, so
might not be a good load for
your amplifier – will cause the
amplifier to overheat/distort.

Speaker Enclosures
Baffles
A speaker suspended by itself would interfere with its own input. As the speaker cone
moves, it pushes air from both the front and the rear, each wave produced being the
same, but 180 degrees out of phase. Therefore they can cancel one another out,
especially at low frequencies. To prevent this cancellation, the front and rear waves must
be separated. This is done by using a baffle, defined as a device which routes the sound
energy. There are many different baffle configurations, called Direct Radiators. A direct
radiator is a speaker which propagates sound directly into the listening area i.e. there are
no acoustic elements between the speaker and the ear, except air!

Infinite Baffle
Theoretically an infinite baffle would be a speaker mounted on a board of infinite length
and width, so that the rear wave produced would never meet with the front and no
cancellation would occur. An example of this would be to mount a speaker in a wall
between 2 rooms. Sound would go into one room for the front wave (the front of the cone
of the loudspeaker), and sound would go into the other room from the rear wave (the back
of the driver). This would not be a very practical design though!

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One way around this is to place the speaker in a sealed box, so there is no way for the
rear wave to get out, therefore no cancellation. The rear wave gets compressed and acts
like a spring pushing the cone forward, making the speaker inefficient and it reduces the
sound pressure level because half of the energy is wasted.

Fig 7. Infinite Baffle Enclosure

Finite Baffle
To increase the efficiency of the loudspeaker would be to open up the back part of the
cabinet. This allows the speaker to move more freely and increase the sound pressure
level. Depending on the size of the opening, there will be a certain amount of phase
cancellation, (front wave interacting with the rear wave).

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Fig 8. Finite Baffle Enclosure

Bass Reflex
All drivers (loudspeakers) have a certain frequency response, and within that response
there is a free air resonant peak at low frequencies (the resonant frequency of something,
is the frequency it vibrates at naturally when excited by a stimulus). This can cause
boominess.

Fig 9. Air Resonances

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By having vents of different sizes and positions, we can eliminate boominess and get a
more efficient loudspeaker. A bass reflex enclosure uses these principles and usually has
its ports at the front to give a longer time for the wavelength to travel before exiting the
enclosure (longer wavelengths for lower frequencies).

Fig 9. Bass Reflex Enclosure

The bass reflex cabinet has to be “tuned” because of its resonant frequencies, so
therefore can be huge to obtain proper phase inversion (i.e. the wavelength is long for low
frequencies [3.4m for 100Hz] so needs a long path before exiting the cabinet). We can
use a vent with a duct to reduce the size of the cabinet whilst maintaining proper tuning.

Ducted Port
Ducted ports have a tube before the port hole in order to select the proper resonant
frequency of the cabinet

Shelved Port
A shelved port has a “shelf” which extends back into the cabinet, allowing wave alignment,
as well as a larger port area.

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Acoustical Labyrinth
This is a shelved and ported cabinet with an extra vertical extension for the sound to travel
through. This places more load on the rear surface of the speaker cushioning the sound
and giving the cabinet a smoother, equal response.

Fig.10 Ducted Port, Shelved Port and Acoustical Labyrinth

Horn Loaded
A horn loaded enclosure is where the driver is mounted at the back of a horn. They are
the oldest type of baffling dating back to the days of Edison and the phonograph.

The basic function of the horn is to couple the driver to the body of air that is in the horn
this provides a form of impedance matching. A horn has high acoustic impedance at the
driver, (the throat of the horn), which has low air particle movement and, this is connected
to the lower or normal acoustic impedance and particle movement air at the mouth of the
horn. This allows the driver diaphragm or cone to be much smaller and move within a
small excursion – allowing a more accurate reproduction of the waveform whilst providing
substantial output. This increases efficiency of the driver.

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Fig 11. Horn

A driver diaphragm should be larger than the throat of the horn, which forces higher
pressure into a small area. This diffracts the sound into the horn. When this is done, there
must be a pressure applied to the rear of the driver or the speaker will move in more than
it will move out. This can cause non-linear distortion (making the movement
asymmetrical). Compression drivers are used to compensate for this, equalising the
pressure of the diaphragm on both sides to prevent non-linear distortion.

Fig 12. Horn with nonlinear motion of the diaphragm

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Fig 13. Horn with linear motion of the diaphragm – a compression driver

Horns can be used for woofers (low frequencies) or tweeters (high frequencies). The
properties of the, flare and mouth of the horn will determine its frequency range, and
dispersion abilities.

Front and rear horn loading combines horns and bass reflex enclosures with the same
driver, achieving the best of both worlds. A front loaded port means the front wave is
projected by the horn and the rear wave is projected by a port. Rear loading is the exact
opposite (see Fig 14. right picture).

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Fig 14. Front Loaded Horn(left) and Rear Loaded Horn (right)

For low frequencies, the horn needs to be very long, and these are achieved by having a
folded horn. There are various designs to achieve this.

Fig 15. Folded Horn

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Crossovers
Most speakers have more than one size driver. This is because it is near to impossible to
get one size of driver to cover the whole audio spectrum clearly and efficiently and without
distortion. By having drivers of different sizes in an enclosure, we can let each driver
supply their best frequency ranges. To do this, we require a crossover. A crossover splits
the audio spectrum into specific frequency bands to be sent to specific drivers, which are
optimized or efficient at handling the frequencies in these bands.

Crossover Classification
There are two types of crossover:
• Passive crossovers, (also know as a high level crossover)
• Active crossovers (also know as electronic crossovers).

Passive Crossovers
A passive crossover uses an inductor and a capacitor. An inductor (usually a coil of wire
around a rod of iron) rises in impedance as frequency rises therefore it has a lower
impedance (lower resistance) for lower frequencies. It acts to block high frequencies and
let low frequencies through and is essentially a low pass filter. A capacitor is an electronic
device that has lower impedance for higher frequencies. It acts to block low frequencies
and let high frequencies through and is essentially a high pass filter. Usually a resistor will
follow a capacitor to decrease the amount of power going to the high frequency driver, as
the high frequency driver (tweeter) will usually require less power than the low frequency
driver (woofer).

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Fig 16. Crossover components

A passive crossover is housed in the loudspeaker enclosure, taking the high level current
supplied by the amplifier. This is why they’re also called “high-level” crossovers. Because
they are hidden away inside the speaker enclosure the filter cut off point is usually fixed.

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Fig 17. Passive Crossover

Active Crossovers
Active crossovers are placed before the amplifier, so work at low voltage levels or line
level and therefore require their own power supply. The signal is split into frequency
sections as before, and each frequency band is then sent to separate amplifiers.

Fig 18. Active Crossover

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Passive vs. Active
For the lower budget PA systems, passive crossovers tend to be cheaper and easier to
use, as the crossover frequency is set.
Active crossovers tend to be more efficient and have more flexibility, as you can change
the crossover point and the level of that frequency band giving more control over the
sound, and giving dedicated amplifiers for the specified frequency range.

Level Reduction And Crossover Frequency

The level reduction at the crossover frequency is usually 3 - 6dB. Each side of the
crossover will then reduce at a certain rate of decibels (dBs) per octave know as the filter
slope. Crossovers are classified in 5 “orders” depending on the filter slope they implement.
1st order crossovers implement a 6dB per octave filter slope. This means that if the cut off
frequency of a filter is 100hz then the signal one octave below 100hz, i.e. 50hz will be 6dB
lower in level.
2nd order crossovers implement a 12dB filter slope.
3rd order crossovers implement an 18dB filter slope
4th order crossovers implement an 24dB filter slope
5th order crossovers implement an 30dB filter slope although these are very rare due to the
cost involved in the manufacture. Lower budget crossovers are usually 1st order.
Crossovers can also be constructed using mixed order filters. For example a crossover
with a 2nd order LPF and a 3rd order HPF. The filter cut off points of the HPF and LPFs are
usually set above or below the actual crossover point to give a slight level reduction at the
crossover frequency. This is to allow for the acoustic coupling that will naturally occur
giving a 3-6dB boost where two phase coherent drivers in close proximity are producing
the same frequency at similar levels, see fig 19 below.

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 st
Fig 19. Crossover graph for a 1 order crossover

DSP crossovers
DSP – Digital Signal Processing has been used in crossover technology for the past 15
years. This enables more accurate crossover filter alignment. With greater EQ power to
shape individual driver responses (the different drivers for the low and high frequencies)
and steeper crossover filter slopes for improvement of directional (in front of the
loudspeaker) and off axis (a determined angle from the centre of the loudspeaker)
performance of loudspeakers are much easier to construct with DSP technology.

Amplifiers
The purpose of an amplifier is to produce an output signal as closely identical as possible
to the input signal, but at a higher level (therefore making it louder).

It can be very confusing reading and understanding a spec sheet of an amplifier Let’s take
a look at the specifications of an amplifier, and explain what they all mean so you can
chose your amplifier wisely!

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Power Output
Power is a measurement of how much work the amplifier can do, but not all the work the
amplifier does will be output as sound, some of it’s output will inevitably be as heat.

Which is louder a 1000 Watt amp or a 1200watt amp?

Well it all depends on how efficient the amplifier is!

And it also depends on whether we are talking Watts as Average RMS or Peak!

Power can be rated by many different methods, so don’t be caught out! They are all based
on the size or level of the amplified waveform. Fig 20. shows an example of an amplified
waveform, and the ratings.

Fig 20. Sinewave

Measurements Relationship
RMS 0.707 x Peak
RMS 1.11 x Average
Peak 1.414 x RMS
Peak 1.57 x Average
Average 0.637 x Peak
Average 0.9 x RMS
Peak to Peak 2.828 x RMS

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RMS 0.3535 x Peak to Peak

RMS – Root Mean Square is the best method of rating the effective voltage, so it should
be used for determining power.
Power is the output voltage squared and the divided by the impedance of the speaker(s) it
is powering.

P = E2/R
Where: P = Power (Watts (W))
E = Output voltage (Volts (V))
R = Impedance (Ohms (Ω)

Amps Need a Load!

All amps need a load to drive and most amps expect a nominal load for instance 4 or 8
Ohms.
So it is important when you are reading or stating the power output of an amplifier to state
it with a given load impedance. Impedance will also change with frequency. Just because
it may say 800W, doesn’t mean that it will deliver that across the frequency range. Usually
the nominal impedance will be stated, and/or a minimum load impedance. For example an
800W (RMS) amp that expects an 8 ohm load will deliver 800W into 8 Ohms but if a 4
Ohm load is applied it will deliver more like 1600W and if a 16 Ohm load is applied it will
deliver more like 400W.

The ability of an amplifier is affected by the circuit’s impedance according to Ohms Law.
Ohm's law states that the potential difference (or voltage drop) between the ends of a
conductor (for example, a resistor) and the current, flowing through the resistor are
proportional at a given temperature.

V = IR
Where : V = voltage (Volts, V)
I = Current (Amps, A)
R = Impedance (resistance) (Ohms, Ω)

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Working Out Speaker Impedances
If one 4Ohm speaker is connected to an amp a 4 Ohm load is applied. If two or more
4Ohm speakers are connected to an amp the impedance will depend on how they are
wired.

Wired In Series

To work out the impedance of this circuit you would use this formula:
it = i1 + i2 + i3 + i4
(Total Impedance = impedance 1 + impedance 2 + impedance 3 + impedance 4 )
Measured in Ohms (Ω)
So if all speakers had an impedance of 4Ohms:
4Ω+4Ω+4Ω+4Ω= 16Ω

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Wired in Parallel

If 4 x 4 ohm speakers are wired in parallel the load is more difficult to work out, the formula
is as follows:

it = 1
(1/i1 +1/i2 + 1/i3 + 1/i4)
Total impedance = 1/(1/impedance1 +1/impedance2 +1/impedance3 +1/impedance4 )
So if all 4 speakers had an impedance of 4Ohms:
1/4 = 0.25 so
0.25+0.25+0.25+0.25 = 1 so
1/1= 1

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Frequency Response
The frequency response of the amplifier should be as flat as possible across the frequency
spectrum (20Hz – 20KHz). On the specifications, there should be a +/- variance shown,
showing how smooth and accurate the response curve is. Don’t be fooled when you see
5Hz – 30kHz on a spec sheet; the +/- variance could be really big! It is also important to
note the power that the measurement has been taken at. If the amplifier has been
measured at a low level, it is very possible that the frequency response will be more like a
sine wave at high powers than low powers, with big peaks and troughs (see Fig 21).

Fig 21. Frequency response and amplifier level

Total Harmonic Distortion (THD)
When an amplifier is at full power, usually there is a slight tonal change to the output
signal. This is caused by the amplifier producing added harmonics, along with the original
signal. These harmonics are known as ghost tones, or harmonic distortion. The Total
Harmonic Distortion (THD) that the human ear can withstand is about 1%. Anymore than
this, and it is annoying, even though the harmonics are musical (whole number integers –
e.g. 110Hz or 1000Hz [not 110.2Hz or 1001.3Hz])

Inter-modulation Distortion (IMD)

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These are similar to the ghost tones with THD, but they are not musical. They are
produced by the sums and differences of two frequencies (e.g. 100+230Hz or 400-210Hz).
It is more annoying that THD, and should be as little as possible.

Slew Rate
Slew rate refers to how fast the amplifier responds to the input signal. The measurement is
similar to 2 motorbikes being tested for their acceleration. If bike 1 takes 3.5 seconds to
get to 60 mph, and bike 2 takes 2.5 seconds to get to 60mph, then bike 2 obviously has
the greater acceleration.

It is argued that there isn’t an upper limit/overkill for a slew rate, but the minimum accepted
slew rate is calculated with the following formula:

2 x π x F x PVO/1,000,000
Where: π = pi (3,1416)
F = test frequency (Hz)
PVO =peak voltage output of the amplifier

Signal to Noise Ratio

Every piece of electronic equipment has inherent noise called thermal or Gaussian noise.
It is impossible to get rid of this noise, but it is possible to have the musical signal large
enough to make the noise unnoticeable. The minimum recommended signal to noise ratio
is 65dB.

Input Sensitivity
The input sensitivity is the minimum input signal level required to give a specified output
level. Many amplifiers have a control to increase the input sensitivity. They do affect the
loudness of the amplifier, but only by adjusting the level of the incoming signal.

Damping Factor
The damping factor refers to the ability of the amplifier to control the speaker’s
movements. The higher the damping factor, the greater control the amp has over the

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speaker, and therefore has a more accurate response. The damping factor should be
over 20 to be acceptable.
To calculate the damping factor, we can use the following equation:

DF = Zload/Zsource
Where: DF = damping factor
Zload = load impedance (Ohms, Ω)
Zsource = source impedance [sum of the internal output impedance of the
amplifier] (Ohms, Ω)

Matching amplifiers with speakers

Apart from making sure the impedance of the amp and speakers is matched we much
match the power (Watts).

If we have an 8 Ohm speaker cabinet that has the following power rating:

Peak8888.600W
Program8......300W
Continuous.....150W

What kind of power amp should we power it with?

We need to know the definitions of the different ratings. Note that each is twice that of the
previous.

Peak: means that maximum peak power the speaker can handle without over-heating the
voice coil. We are talking very short transient peaks of around 10 milliseconds.

Continuous: As opposed to peak this means a continuous signal at a constant level

something like a sign wave or test tone over the long term. Again we are looking at what
the speaker can handle without over-heating the voice coil.

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Program: This means the maximum normal load over the medium term. This is designed
to test the speaker’s capability of a varying dynamic signal similar to that of musical
program.

So, what kind of power amp should we power it with?

Something that has an output of 600W into 8 Ohms? Well you might think this because it
is a typical misunderstanding which comes from home HiFi speaker matching where more
headroom in the amp is required for musical peaks. This is ok for lower listening levels but
things change as soon as we get to the higher SPL required of larger PA systems.

For instance: In our example our 600W 8 Ohms amp is most likely to be RMS rated so
when working with normal dynamic musical program it will be working around 300W in to 8
Ohms (around 3dB below its max RMS level) but it may well peak at 3dB above that level
meaning the speaker could be asked to handle a 1200Watt peak transient. This would of
course blow our 600W 8 Ohm speaker.

The better matching would be something around 280Watt in to 8 Ohms this would mean
that if the amp peaks at 3dB (double the wattage) above this (560Watts) it wont blow the
speaker which can handle a max peak transient of 600Watt.

Headroom
Headroom is the reserve power needed for peaks in the program material. A high quality
sound system would usually require up to 10dB of headroom. Therefore if we have a
200W amplifier, most of the time it is only delivering 2-20W (10dB is equal to ten times the
power). The remaining power is just for the peaks in the material being played. If an amp
is clipping (distorting the signal) all the time, it simply isn’t powerful enough and needs to
be bigger, or the speakers will be damaged.

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