Download 1 Components of a medium size PA

i. Components of a medium size PA
Contents
a.
b.
c.
d.
e.
f.
g.
h.
Connectors: i. XLR,
ii. Jack,
iii. Phono
iv. Speakon
Cables: i. Speaker,
ii. Signal,
iii. Microphone
iv. power
Signal sources: i. Microphones,
ii. Instruments/Direct injections boxes,
iii. Playback
Sound mixers: i. Microphone input
ii. Line input
iii. Controls
iv. Busses
v. Outputs
Amplifiers: i. Signal input
ii. Speaker output
Graphic equalisers: i. Signal input
ii. Signal output
iii. Real time analyser
Electronic crossovers: i. controls
ii. Signal input
iii. Signal output
Main PA speaker systems: i. Loud Speakers
1. Single box full range
2. Active Multi enclosure
3. Loudspeaker parameters
a. Sensitivity
b. Frequency response
c. Power handling
i. Thermal
ii. Mechanical
iii. Fatigue bending
4.
4. Power compression
5. Directivity
6. Impedance
Loud speaker input connections
a. Speacon
i. 2 pole
ii. 4 pole
iii. 8 pole
iv. Xlr 3 pole
1. Components of a medium size PA
a. Connectors: - There are 4 common types of audio connector used
in PA systems today
i. XLR,
The 3 pole XLR connector sometimes known
as a Cannon, due to the Hover effect, ITT
Cannon were amongst the first manufacturers
to make this type of plug. Today this is
generally used to connect balanced
microphones to the mixing desk microphone
inputs. They are also used to connect line
level balanced outputs of mixing desks to
crossovers line level balanced inputs and
crossovers line level balanced outputs to
amplifier line level balanced inputs . Pin 1 is
always used to connect the screen, pin2 and
pin 3 are used to connect the positive, in
phase or hot signal wire and the negative, out
of phase or cold signal wire, there is no
universal standard for the connection of pins
2 or 3, you just have to look in the desk
manual and at the microphone data to work
out if positive sound pressure will produce a positive signal at
pin 2 or pin 3.
ii. Jack plugs
Jack plugs come in two types, two-pole and three poles
The two-pole type is only used in un
balanced situations, for example for line
level inputs on mixing desks, on instrument
amplifiers or guitars. This type of plug
should only be considered for use when a
signal does not have to travel far from its
source to its destination, as an un balanced
connection dose not have any inherent
immunity to magnetic interference
The three-pole type can be used for connection
of head phones, in Insert loops on mixing
desks, tip is usually send ring is return and
sleeve is screen. It can also be used in
balanced situations, for example for line level
inputs on mixing desks which has a balanced
line 3 pole jack input, or on a piece of
equipment which has a balanced line 3 pole
jack output, tip is usually in phase signal ring is out of phase signal and
sleeve is screen.
iii. Phono plugs
Phono plugs sometimes known as RCA
Phonos are also only two pole devices so
the same sort of restriction on their use as
the jack plug must be applied, it is un
balanced. This type of plug is most
commonly found on consumer grade HiFi
i.e. CD players, Tape decks, Mini disks,
centre pole is usually the in phase signal
and the sleeve is screen.
iv. Speakon loudspeaker connector
Speakon loudspeaker connector come in three
types the most common is the 4 pole, which is
normally used to connect one speaker circuit down
a 2 conductor cable. Pole 1+ is used for the
positive connection and 1- is used for the negative
connection. However in a more professional
environment the 4 pole plug can be used connect
two speaker circuits down a 4 conductor cable.
Poles 1+ and 1- are usually used for the lower frequency speaker of the
pair and pole 2+ and 2- are used for the higher frequency speaker, but be
careful because no universal standard exists, it is down to the system
designer. There are also a 2 Pole version which mates with a 4 pole socket
but only with 1+ and 1- and there is an 8 pole variant capable of
connecting up to 4 speaker circuits.
NL4MMX Speakon loudspeaker connector 4 pole
To 4 pole : two speaker circuit
NL2FC Speakon loudspeaker connector 2 pole
one speaker circuit
NL8FC Speakon loudspeaker connector 8 pole
Four speaker circuits
b. Cables: - All the cables used in PA systems fall into three
categories.
i. Speaker cables
These are used to connect power amplification equipment to the loud
speakers. They do not have an electrostatic screen. The most common
Speaker cables use two signal conductors per circuit one for the in phase
signal and one for the 0volts or ground.
Multi
core speaker cables are available for use with 4 pole and 8 pole connectors
which provide 2 and 4 circuits respectively, the advantage of these is that
you need to layout less individual cable runs. The cables should have the
biggest conductor cross sectional area physically possible, to provide the
least resistance and as a result the most efficient transfer of the power from
the power amplifier to the speaker. It is preferable that the conductor be
made up from multiple fine strands, which makes the cable more flexible
and increases the surface area of the conductor, which helps reduce high
frequency skin effects. Optimally speaker cables should only be just long
enough to reach from the amplifier to the speaker again to reduce
resistance
The voltage present in a speaker cable varies from a few volts to 10s of
volts. The current flowing in a speaker cable can run into 10s of amps
cable.
ii. Signal cables,
These cables connect signal sources to amplification equipment. These
always have an electrostatic screen to reduce the interconnections
susceptibility to electrostatic interference. This screen may be made of a
Lapp winding of fine strands of wire laid side by side or an interwoven
layer of fine strands, which make a jacket around the inner conductors.
Modern signal cables often employ a foil screen coupled with a drain wire
made of a few strands of fine wire. This drain wire allows connection of
the foil screen to the connector terminal. Signal cables are subdivided into
two types.
The simplest are for use with unbalanced signal interconnections, they use
only one signal conductor and a screen per signal.
These are generally only used for
short distances, for example from
guitars to guitar amps and from desk
aux sends to effects units inputs and
back to the aux returns.
The more common are balanced line cables and have two signal
conductors and a screen; the two signal conductors are twisted together to
reduce pickup of electromagnetic interference.
To enable lots of signals to be connected from the stage microphones and
Di’s to the mixing desk inputs and from the desk to the amplifiers on the
stage, a cable known as a multicore is used.
This is basically multiples of
microphone twisted pair screened cables in a single outer sheath.
The voltage present in a signal cable varies from a few 1000ths of a volt in
a mike cable, to several volts in a hard driven line level circuit, used
between desks and amps and desks and effects units. There is only a very
small current flowing in any signal cable.
iii. Microphone cables
Microphone cables have two signal conductors and a screen; the two
signal conductors are twisted together to reduce pickup of electromagnetic
interference
iv. Mains power cables
These are used to connect mains power to amplification equipment. They
do not have an electrostatic screen. The most common Mains cables use
three conductors per circuit, one for Earth (green/yellow) one for live
(brown) and one for neutral (blue).
Where more power is required 3 phase power may be used these cables
use five conductors per circuit, one for Earth (green/yellow) three for the
lives and one for neutral the colour for the lives and neutrals vary so check
what is being used for what
Mains cables are available in lots of cross sectional areas, each capable of
passing a particular maximum current, the bigger the cross sectional area
of a cable, the larger the current which can be carried by it.
To prevent the cable from being overloaded it must be protected by a
suitably rated fuse or circuit breaker. It is preferable that the conductor be
made up from multiple fine strands, which makes the cable more flexible.
Mains cables should only be just long enough to reach from the power
supply outlet to the amplification equipment and any slack should not be
tightly coiled up as this can result in the current flowing through the cable,
causing heat that cannot escape resulting in the coiled wire melting or
catching fire.
The voltage present in a mains cable varies from 240 volts in a singlephase circuit to 415 volts, in a 3-phase system. Obviously 415volts is more
dangerous than 240 volts, but both are potentially lethal. The current
flowing in a mains cable can run into 1000s of amps. and must always be
treated with respect. If there are any faults with a mains cable do not use
it!!
All cables must be correctly rated and specified for the use they are going
to be put to.
c. Signal sources: i. Microphones,
Microphones are the interface between any performer and the PA system;
their function is to convert tiny vibrations in the air, sound waves, into
electrical power. This power can then be electrically amplified to a level
sufficient to drive loud speakers, which are microphones in reveres
converting electrical power into sound. From the users point of view the
frequency response, dynamic range and the polar response are what
determines what jobs a microphone will be good at. Generally in PA
applications a frequency response of 50 Hz to 15 kHz is usable and a
cardiod polar response is best, as it rejects sounds which are not coming
from the direction the microphone is pointed.
There are three main methods that microphones use to convert sound
into electricity.
1. Dynamic microphones
The most common are known as dynamic microphones, these are
simply a coil of wire
coupled to a diaphragm that
is free to move in response
to sound waves hitting it.
The coil is suspended in a
powerful magnetic field,
which induces a current to
flow in the coil when the
sound wave hitting the
diaphragm causes it to
move in the magnetic field.
2. Condenser
Another popular type of microphone is the condenser, this uses a pair
of small foil plates one fixed to the body of the microphone and the
other free to move in response to sound waves, using a small piece of
electronics a charge is placed on the plates and the resulting voltage
across the plates measured by the electronics, when the distance
between the plates is caused to change by the sound wave, the voltage
across the plates changes proportionally. This signal suitably amplified
within the microphone can then be used as the output signal. Power to
enable the “on board electronics” to work, in the case of a true
condenser microphone is
provided by phantom power
from the mixing desk, however
some very similar microphones
known as electret microphones
employ an AA cell battery
housed within the microphones
body to provide the necessary
power.
3. Ribbon microphones
There are still a few examples of ribbon microphones about, these use a
thin ribbon of conductive foil suspended in a strong magnetic field. The
ribbon acts as both diaphragm and coil from the dynamic description
above. When sound hits the ribbon the movement of the ribbon in the
magnetic field causes a current to be induced in it. This current is fed to
a transformer that transforms the relatively low voltage high current to a
higher voltage low current form which is suitable to connect to a mixing
desk microphone input
ii. Instruments/Direct injections boxes,
Most musical
instruments do not have
a low impedance
balanced line type
output.
In the case of most bass
guitars, they have a high
impedance output.
This means that they are
unsuitable to send down
long signal cables.
The best way to connect
the output of a bass,
acoustic guitar, keyboard or sub-mixer to a microphone input of a mixing
desk is via a DI box (Direct Injection box). This device converts an
unbalanced high impedance output from a bass guitar pickup into a
balanced line low impedance source. It also permits the instrument to be
connected to the back line amp via the “through” connection on the Di
Box .It also enables any earth loop between the instrument end and the PA
to be broken, preventing earth loop hums . Note some Di Boxes are
passive and some are phantom powered requiring 48-volt phantom power
iii. Playback
The output level from a device of this type is known as line level and is of
the order of 1 volt.
If a play back device such as a CD player is to be used remote to the
mixing desk and assuming it dose not have its own balanced line output,
then DI boxes should be used to interface it with the mixing desks
microphone input. If the device is local to the desk it can be connected
using a short (2 m long) unbalanced Phono to jack signal lead and
connected to the desks line input
d. Sound mixers are the next link in the audio chain, they come in a
vast range of variants, but all perform a similar task. All have the
following
i. Microphone input
. Generally the microphone inputs are made through 3 pole XLR
connectors and are balanced.
ii. Line input
line inputs are usually made via jack connectors which are sometimes
balanced and use 3 pole plugs or unbalanced and use 2 pole plugs
iii. Controls
Depending on the particular desk you are using, the degree of control
available varies, but basically all desks channels have an input gain
control, a tone control section, auxiliary sends, a pan and a master channel
fader
1. Input gain
The function of this control is to allow the input signal level to be
matched to the signal level required for the electronics inside the
desk to operate optimally, so as not to cause it to overload or so
that the signal in the electronics is so small that the electronics
own background noise is big relative to the signal. The gain
control provides this function by, as you might expect, adjusting
the level of amplification applied to the incoming signal as soon
as it enters the desk. Most mixing desks have an attenuating (pad
–20 dB) switch which allow the microphone input sensitivity to
be reduced, to prevent input overload. The other function that is
controlled in this area of this particular desk is the phantom
power to enable provision of power for phantom powered input
devices
2. Tone controls
On a very simple desk there will be at least two tone controls,
these will control a band of frequencies known as bass, which
extend from 20 Hz to 200Hz and a band of frequencies known as
treble which extend from 5kHz to 20kHz. . The rang of the
controls are of the order of +- 15 dB. The operation of these two
controls will be very similar to that of treble and bass on a home
Hi Fi.
two band fixed tone control
On a slightly more sophisticated desk there will be three tone
controls, these will control a band of frequencies known as bass
which extend up to 200Hz, a band of frequencies known as mid
which covers the range from 200 Hz to 5kHz and a band of
frequencies known as treble which extend from 5kHz up. The
operation of these controls will be similar to that of treble and
bass on a home Hi Fi but the mid control is a has a notch
characteristic where as the bass and treble are shelving, however
we will still class this three band tone control as simple as its
effects are relatively board and not particularly specific, like
using a shovel in comparison to a fine knife.
three band fixed tone control
The next level of sophistication is the inclusion of a frequency
sweep facility with the mid control, the width of the frequency
band that this type of control has, is much narrower than the
simple mid control, already mentioned. There is a further control
knob associated with the level control knob of a swept mid filter,
this control knob is the frequency sweep, it determines the centre
frequency of the filter, i.e. it points at the frequency in the middle
of the band it will allow the mid level control to adjust. So lets
say you want to adjust the level of the frequencies in the area
around 3 kHz you would adjust the mid sweep control to point at
3 kHz and then use the mid level control knob to adjust the level
of the band of frequencies in that region. This type of control is
much more precise than the simple mid tone control as it lets you
control a selected small range of the audio spectrum rather than
the broad brush effect of the simple mid.
three band fixed bass and treble with swept mid tone control
The most precise control is provided by a tone control known as
a swept mid with variable “Q”, ”Q” is a term used to define the
width of a filter, the higher the “Q” figure the narrower the band
of frequencies affected by the filter conversely the lower the “Q”
the wider the band affected. High “Q” is sometimes called peaky,
so if there is a particular frequency that is causing a problem the
filter can be adjusted to just affect that area of the spectrum and
not detract from areas, which are not.
three band fixed bass and treble with swept midand variable
“q" tone control
There is one other part to the tone control known as a high pass
filter. On simple desks this will be a switch which when pressed
cuts out any sound below the cut-off frequency of the filter.
Normally fixed high pass filters have a cut-off of between 80 Hz
and 120 Hz.
On more professional mixers this control has a variable cut-off
frequency, again this improves the use of the filter as it can be
arranged only to remove the precise part of the audio spectrum
required. The purpose of the high pass filter is to remove sub
sonic frequencies or frequencies in a signal below its useful range
It is worth noting that the adjustment of tone controls can affect
the required setting of the gain control to provide optimal
matching of the input signal.
The most sophisticated tone control incorporates variable
frequencies and “q” on all its bands
iv.
sub mix busses
From the mixers input channels, the signal is fed to the desks sub
mix busses to enable multiple channels to be grouped together
before being sent to the desks master so the whole of a section eg
drums or backing vocals can be controlled together .
v. Outputs
From the mixers input channels and sub mix busses, via the
channel master faders, the signal is fed to the desks main PA
summing stage, here all the outputs from the separate channels
are added together and then fed to the mix main output sockets
via the mix master fader. In the same way the auxiliary sends are
sent to there respective aux master summing stages, here all the
aux outputs from the separate channels are added together and
then fed to there respective aux master output sockets via there
respective aux master control knob. To prevent internal
overloading of the mixer before sufficient signal level is
produced at the output connector it is always sensible to work
with the output master sand aux master control knobs set to 0dB
i.e. the master control is not boosting or attenuating the signal in
the desk.
e. Amplifiers: The purpose of a power amplifier is to amplify the line level signal coming
from the output of the mixing desk or the system controller to a
sufficiently large level that it will drive loudspeakers to the volume
required
i. Signal input
There are a few different input level requirements for power amplifiers
depending on the particular unit you are using, some require only 750 mV
input signal present to produce their full rated output power, some 1V and
some others 1.4 V what is important is that the piece of equipment feeding
the input signal to the amplifier is capable of driving the necessary signal
level without distorting. There a several different input connections used
on power amplifiers,
the most desirable is a balanced line XLR, however, unfortunately, many
semi pro units only have unbalanced jacks that generally result in earth
loops and hum problems.
ii. Speaker output
The output power that is available from an amplifier and the impedance it
is capable of driving is depend on its design. The rating of an amplifier
will usually define the maximum output power into the minimum
impedance that the amplifier can safely drive, commonly the minimum
impedance will be between 2 ohms and 16 ohms .If a speaker system with
a lower impedance than the amplifiers minimum is connected to this
amplifier, then the speaker will try to pull more current from the amplifier
than the amplifier can provide and the amplifier will either current limit or
may over heat , fail and go up in
smoke!*****************************************. It is essential
that any speaker system connected to an amplifiers output is capable of
handling the maximum output power of the amplifier at the impedance of
the connected speaker system, preferably with a factor of safety to improve
the life expectancy of the speaker.
If a speaker of lower power handling capacity, for example a 50 watt rms 8
ohm hf driver is connected to a 200watt 8 ohm amplifier then the amplifier
must be limited by a system controller/ active crossover to a suitably low
level ie 50 watts into 8 ohm(it gets complicated)
f. Graphic equalisers: This is a convenient series of tone controls which can be used provide
overall tone control of a signal/system. Thy are arranged as a series of
faders side by side with each fader controlling a fixed band of frequencies.
The mid point of the faders normally is labelled 0dB indicating that in this
position the fader is neither boosting nor cutting the band that it controls.
The
control
rang of the
faders are
+- 15 dB
which is
fairly
large.
Sometimes a switch which reduces the rang to +- 6 dB is provided so that
more resolution in setting can be obtained, but only over half the range.
Some graphics also have a high pass filter similar to those fond on a
mixing desk bur with their cutoff frequency fixed at about 40 Hz
i. Signal input
The input to a graphic is usually at line level via Jack plugs or XLR’s,
most of the better units employ balanced XLRs
ii. Signal output
The output from a graphic is usually at line level via Jack plugs or XLRs,
most of the better units employ balanced XLRs
iii. Real time analyser
A few graphics also incorporate a real time analyser to assist setting up the
graphic .To do this a test signal, usually pink noise must be applied to the
system and a calibrated microphone connected to the “ Real time analyser
“
g. Electronic crossovers: Electronic crossovers do the same job as passive crossovers in loud
speakers; they divide the audio spectrum up into a series of bands, in the
simplest case bass and treble. The purpose of this division is to send the
correct signal frequencies to the rite type of speaker to reproduce it, i.e.
bass to the bass driver and treble to the tweeter etcetera. The difference
between active and passive crossovers is where they divide the signal.
Actives do it at a point in the signal chain between the output of the
mixing desk and the input to the power amplifier at line level, and passives
do it at a point between the amplifier and speakers drivers at speaker
driving level.
i. Controls
On a very simple active crossover there may be no controls at all if it has
been designed to work with a specific amplifier and speaker system, but on
a unit designed to be flexible the normal controls permit adjustment of the
crossover frequencies and the level of each bands output. More specialist
units also permit control of time alignment of the bands output to alow the
sounds reproduced by the various speakers to arrive at the listeners ear at
the same time , although each transducer in the speaker system may be at a
different distance from the listener. Control of the maximum signal level
sent to the amplifier and thus protection of the speakers may be provided
by this device
ii. Signal input
The input to a Electronic crossovers is usually at line level via Jack plugs
or XLRs, most of the better units employ balanced XLRs
iii. Signal output
The output from a Electronic crossovers is usually at line level via Jack
plugs or XLR’s, most of the better units employ balanced XLRs
h. Main PA speaker systems: Depending on what type of event the equipment is being used for, the
quantity of speakers will vary from one to hundreds and the power used to
drive them from a few watts to hundreds of thousands of watts. The type
of enclosure varies and the number of transducers in each cabinet will also
vary from a single full range unit, to a series of frequency specific drivers.
A perfect speaker would be able to reproduce sound from 20 Hz to 20 kHz
and have a very well defined dispersion pattern which is independent of
frequency
i. Loud Speakers
.
1. Single box full range
A Single box full range system is the simplest and tend to be enclosures
with two speaker chassis, one to handle bass frequencies up to about
3.5kHz and a small tweeter to handle the frequencies from here up to 16 to
20 kHz. The bass unit is fed from a passive crossover with signal between
40 Hz and the frequency at which the tweeter comes in at and the tweeter
fed with high frequency signal at which the bass unit is crossed out. The
main problem with these is the large bass unit must produce relatively high
frequencies, which for its size and its mass is not ideal and its coverage
angle will tend to be poor at these higher frequencies resulting in beaming.
I.e. you stand directly in front of the cabinet on its axis and it sounds loud,
but as you move of to one side the higher mid drops off, intelligibility is
lost, dose not make for a good uniform coverage. With only a small
tweeter it is not possible to reduce the crossover frequency and thus reduce
the beaming problem because the small tweeter cannot reproduce sounds
lower than its cut-off frequency which is determined by its horn size, the
larger the horn the lower the cut off frequency. Also to handle lower
frequencies the unit would have to be able to handle more power, as it
would be handling more of the audio spectrum, generally power handling
involves extra cost. The reason why this arrangement is popular and
common is the cost of producing such a cabinet is relatively low as the
However if performance is more important than cost, a larger tweeter with
a lower crossover frequency can be employed to reduce the beaming
problem as the bass unit will not have to try to reproduce such high
frequencies any more. These tweeters usually use a compression driver
with a diaphragm about 2” in diameter and a throat 1” in diameter .The
crossover frequency with this arrangement is generally around 1.5 kHz.
Note this is still a bit on the high side for a 12” or 15” bass driver to
reproduce without beaming a bit, but it is considerably better than the
small tweeter in the first case. There are a few systems around which use
an even bigger horn tweeter that is capable of running from 800 Hz. These
tend to be very expensive and heavy, they usually use a compression
driver with a diaphragm about 4” in diameter and a throat 2” in diameter.
Unfortunately the top end frequency response of these larger horns is more
limited than the small ones so the large horn gets rid of the beaming
problem but tends to end up not being able to produce the higher
frequencies very well.
From two driver enclosures the next step is to add a third, which is specific
to covering the mid range that the bass driver is too large to cover and that
the tweeter is too small to cover. This mid driver must be able to handle
the power. Be fast enough to accurately reproduce the transients in the
high frequency mid content of the band. The sensitivity must be high
enough not to limit the sensitivity of the whole system. The type of
transducer can be either cone (like a bass driver only much smaller) or a
large horn with either a large compression driver or cone driver driving it
2. Active Multi enclosure systems
Active Multi enclosure systems. For larger systems the speakers may are
split into sections with the enclosures covering low bass mounted on the
floor and the other units covering bass mid and top raised up above the
audience to improve coverage. There is a second advantage in splitting the
speaker enclosures up so one type covers top one covers mid and so on in
that the overall system can be tailored to suit the event. The splitting up
also reduces the individual cabinet’s weight i.e. if a full range enclosure
with three drivers weighed 120 k a split version with it split into three
cabinets
3. Loudspeaker parameters
a. Sensitivity
Sensitivity figure is a measure of how much sound is produced, at a
particular distance from the speaker, by a particular amount of electrical
power being applied to the speaker. Depending on the particular method a
manufacturer uses, the sensitivity may be a peak reading at a particular
frequency, or could more usefully be an average figure for the whole of the
working bandwidth of the speaker. Normally the distance between the
speaker and the measurement microphone is 1 meter. There is a question if
this distance is measured from the cone or the front of the cabinet as it can
make a significant difference to the measurement .The power applied is 1
watt RMS . Good sensitivity figures should be between 95 and 100 dB
for a bass cone driver 98 to 108 dB for a compression driver and around
100 to 105 dB for a small tweeter
b. Frequency response
The required frequency response of the speaker system depends on its
application. For a full range cabinet 50 Hz to 16 kHz is not bad, but the
nearer to 35 Hz to 40 KHz is preferable. For a bass only cabinet 40 Hz to
200 Hz is not bad, but the nearer to 30 Hz to 200 Hz is better. For mid/top
only cabinet 150 Hz to 16 kHz is ok
c. Power handling
Three things limit a speakers’ power handling.
i. Thermal
The thermal dissipation capability of the voice coil, magnet assembly
and chassis to remove heat from the voice coil
ii. Mechanical
The mechanical strength of the glues and other component parts
iii. Fatigue bending
The ability of the suspension to flex for millions of cycles of
movement of the cone assembly without fracturing through bending
fatigue
d. Power compression
This term defines the effect of the build up of heat in the voice coil, which
causes the voice coil temperature to rise, causing the resistance of the coil
to rise and results in the reduction of acoustic output from the driver for a
given applied voltage i.e. the sensitivity of the driver drops off as the
temperature of the driver rises.
e. Directivity
This characteristic defines the variation of the speakers’ polar horizontal
and vertical frequency response relative to the position of the listener to
the speaker
f. Impedance
The speakers’ impedance is like resistance but varies its load on the
amplifiers’ output with the frequency of the signal applied to it. Normally
it will be specified as a nominal value but is commonly shown in a
graphical form as well.
g. Loud speaker input connections
a. Speacon
i. 2 pole
ii. 4 pole
iii. 8 pole
iv. 3 Pole XLR