AUDIO FREQUENCY RANGE OF ANIMALS:

Hearing range
Logarithmic chart of the hearing ranges of some animals

Hearing range describes the range of frequencies that can be heard by humans or
other animals, though it can also refer to the range of levels. The human range is
commonly given as 20 to 20,000 Hz, although there is considerable variation between
individuals, especially at high frequencies, and a gradual loss of sensitivity to higher
frequencies with age is considered normal. Sensitivity also varies with frequency, as
shown by equal-loudness contours. Routine investigation for hearing loss usually
involves an audiogram which shows threshold levels relative to a normal.
Several animal species are able to hear frequencies well beyond the human hearing
range. Some dolphins and bats, for example, can hear frequencies up to 100,000 Hz.
Elephants can hear sounds at 14–16 Hz, while some whales can hear infrasonic sounds
as low as 7 Hz.

Humans
In humans, sound waves funnel into the ear via the external ear canal and reach
the eardrum (tympanic membrane). The compression and rarefaction of these waves set this thin
membrane in motion, causing sympathetic vibration through the middle ear bones (the ossicles:
malleus, incus, and stapes), the basilar fluid in the cochlea, and the hairs within it, called stereocilia.
These hairs line the cochlea from base to apex, and the part stimulated and the intensity of
stimulation gives an indication of the nature of the sound. Information gathered from the hair cells is
sent via the auditory nerve for processing in the brain.
The commonly stated range of human hearing is 20 to 20,000 Hz.[9][10][note 1] Under ideal laboratory
conditions, humans can hear sound as low as 12 Hz[11] and as high as 28 kHz, though the threshold
increases sharply at 15 kHz in adults, corresponding to the last auditory channel of the cochlea.
[12]
The human auditory system is most sensitive to frequencies between 2,000 and 5,000 Hz.
[13]
Individual hearing range varies according to the general condition of a human's ears and nervous
system. The range shrinks during life,[14] usually beginning at around age of eight with the upper
frequency limit being reduced. Women lose their hearing somewhat less often than men. This is due
to a lot of social and external factors. For example, more than once it has been proven that men
spend more time in noisy places, and this is associated not only with work but also with hobbies and
other things. Women have a sharper hearing loss after menopause. In women, hearing decreases
and is worse at low and partially medium frequencies, while men are more likely to suffer from
hearing loss at high frequencies. [15][16][17]

An audiogram showing typical hearing variation from a standardized norm.

Audiograms of human hearing are produced using an audiometer, which presents different
frequencies to the subject, usually over calibrated headphones, at specified levels. The levels
are weighted with frequency relative to a standard graph known as the minimum audibility curve,
which is intended to represent "normal" hearing. The threshold of hearing is set at around 0 phon on
the equal-loudness contours (i.e. 20 micropascals, approximately the quietest sound a young
healthy human can detect),[18] but is standardised in an ANSI standard to 1 kHz.[19] Standards using
different reference levels, give rise to differences in audiograms. The ASA-1951 standard, for
example, used a level of 16.5 dB SPL (sound pressure level) at 1 kHz, whereas the later ANSI-
1969/ISO-1963 standard uses 6.5 dB SPL, with a 10 dB correction applied for older people.
Cats

Outer ear (pinnae) of a cat

Cats have excellent hearing and can detect an extremely broad range of frequencies. They can hear
higher-pitched sounds than humans or most dogs, detecting frequencies from 55 Hz up to
79 kHz. Cats do not use this ability to hear ultrasound for communication but it is probably important
in hunting, since many species of rodents make ultrasonic calls. Cat hearing is also extremely
sensitive and is among the best of any mammal, being most acute in the range of 500 Hz to
32 kHz. This sensitivity is further enhanced by the cat's large movable outer ears (their pinnae),
which both amplify sounds and help a cat sense the direction from which a noise is coming.

Dogs
The hearing ability of a dog is dependent on breed and age, though the range of hearing is usually
around 67 Hz to 45 kHz. As with humans, some dog breeds' hearing ranges narrow with age, such
as the German shepherd and miniature poodle. When dogs hear a sound, they will move their ears
towards it in order to maximize reception. In order to achieve this, the ears of a dog are controlled by
at least 18 muscles, which allow the ears to tilt and rotate. The ear's shape also allows the sound to
be heard more accurately. Many breeds often have upright and curved ears, which direct and
amplify sounds.
As dogs hear higher frequency sounds than humans, they have a different acoustic perception of the
world. Sounds that seem loud to humans often emit high-frequency tones that can scare away
dogs. Whistles which emit ultrasonic sound, called dog whistles, are used in dog training, as a dog
will respond much better to such levels. In the wild, dogs use their hearing capabilities to hunt and
locate food. Domestic breeds are often used to guard property due to their increased hearing
ability. So-called "Nelson" dog whistles generate sounds at frequencies higher than those audible to
humans but well within the range of a dog's hearing.
Bats
Bats have evolved very sensitive hearing to cope with their nocturnal activity. Their hearing range
varies by species; at the lowest it can be 1 kHz for some species and for other species the highest
reaches up to 200 kHz. Bats that can detect 200 kHz cannot hear very well below 10 kHz. In any
case, the most sensitive range of bat hearing is narrower: about 15 kHz to 90 kHz.
Bats navigate around objects and locate their prey using echolocation. A bat will produce a very
loud, short sound and assess the echo when it bounces back. Bats hunt flying insects; these insects
return a faint echo of the bat's call. The type of insect, how big it is and distance can be determined
by the quality of the echo and time it takes for the echo to rebound. There are two types of
call constant frequency (CF), and frequency modulated (FM) that descend in pitch. Each type
reveals different information; CF is used to detect an object, and FM is used to assess its distance.
The pulses of sound produced by the bat last only a few thousandths of a second; silences between
the calls give time to listen for the information coming back in the form of an echo. Evidence
suggests that bats use the change in pitch of sound produced via the Doppler effect to assess their
flight speed in relation to objects around them. The information regarding size, shape and texture is
built up to form a picture of their surroundings and the location of their prey. Using these factors a
bat can successfully track change in movements and therefore hunt down their prey.
Mice
Mice have large ears in comparison to their bodies. They hear higher frequencies than humans; their
frequency range is 1 kHz to 70 kHz. They do not hear the lower frequencies that humans can; they
communicate using high-frequency noises some of which are inaudible by humans. The distress call
of a young mouse can be produced at 40 kHz. The mice use their ability to produce sounds out of
predators' frequency ranges to alert other mice of danger without exposing themselves, though
notably, cats' hearing range encompasses the mouse's entire vocal range. The squeaks that
humans can hear are lower in frequency and are used by the mouse to make longer distance calls,
as low-frequency sounds can travel farther than high-frequency sounds.

Birds
Hearing is birds' second most important sense and their ears are funnel-shaped to focus sound. The
ears are located slightly behind and below the eyes, and they are covered with soft feathers – the
auriculars – for protection. The shape of a bird's head can also affect its hearing, such as owls,
whose facial discs help direct sound toward their ears.
The hearing range of birds is most sensitive between 1 kHz and 4 kHz, but their full range is roughly
similar to human hearing, with higher or lower limits depending on the bird species. No kind of bird
has been observed to react to ultrasonic sounds, but certain kinds of birds can hear infrasonic
sounds. "Birds are especially sensitive to pitch, tone and rhythm changes and use those variations to
recognize other individual birds, even in a noisy flock. Birds also use different sounds, songs and
calls in different situations, and recognizing the different noises is essential to determine if a call is
warning of a predator, advertising a territorial claim or offering to share food."
"Some birds, most notably oilbirds, also use echolocation, just as bats do. These birds live in caves
and use their rapid chirps and clicks to navigate through dark caves where even sensitive vision may
not be useful enough."
Pigeons can hear infrasound. With the average pigeon being able to hear sounds as low as 0.5 Hz,
they can detect distant storms, earthquakes and even volcanoes. This also help them to navigate.

Insects
Greater wax moths (Galleria mellonella) have the highest recorded sound frequency range that has
been recorded so far. They can hear frequencies up to 300 kHz. This is likely to help them evade
bats.

Fish
Fish have a narrow hearing range compared to most mammals. Goldfish and catfish do possess
a Weberian apparatus and have a wider hearing range than the tuna.

Marine mammals
Dolphins

As aquatic environments have very different physical properties than land environments, there are
differences in how marine mammals hear compared with land mammals. The differences in auditory
systems have led to extensive research on aquatic mammals, specifically on dolphins.
Researchers customarily divide marine mammals into five hearing groups based on their range of
best underwater hearing. (Ketten, 1998): Low-frequency baleen whales like blue whales (7 Hz to
35 kHz); Mid-frequency toothed whales like most dolphins and sperm whales (150 Hz to 160 kHz) ;
High-frequency toothed whales like some dolphins and porpoises (275 Hz to 160 kHz); Seals (50 Hz
to 86 kHz); Fur seals and sea lions (60 Hz to 39 kHz).
The auditory system of a land mammal typically works via the transfer of sound waves through the
ear canals. Ear canals in seals, sea lions, and walruses are similar to those of land mammals and
may function the same way. In whales and dolphins, it is not entirely clear how sound is propagated
to the ear, but some studies strongly suggest that sound is channelled to the ear by tissues in the
area of the lower jaw. One group of whales, the Odontocetes (toothed whales), use echolocation to
determine the position of objects such as prey. The toothed whales are also unusual in that the ears
are separated from the skull and placed well apart, which assists them with localizing sounds, an
important element for echolocation.
Studies have found there to be two different types of cochlea in the dolphin population. Type I has
been found in the Amazon river dolphin and harbour porpoises. These types of dolphin use
extremely high frequency signals for echolocation. Harbour porpoises emit sounds at two bands, one
at 2 kHz and one above 110 kHz. The cochlea in these dolphins is specialised to accommodate
extreme high frequency sounds and is extremely narrow at the base.
Type II cochlea are found primarily in offshore and open water species of whales, such as
the bottlenose dolphin. The sounds produced by bottlenose dolphins are lower in frequency and
range typically between 75 and 150,000 Hz. The higher frequencies in this range are also used for
echolocation and the lower frequencies are commonly associated with social interaction as the
signals travel much farther distances.
Marine mammals use vocalisations in many different ways. Dolphins communicate via clicks and
whistles, and whales use low-frequency moans or pulse signals. Each signal varies in terms of
frequency and different signals are used to communicate different aspects. In dolphins, echolocation
is used in order to detect and characterize objects and whistles are used in sociable herds as
identification and communication devices.

Audio electronics
Audio electronics is the implementation of electronic circuit designs to perform conversions
of sound/pressure wave signals to electrical signals, or vice versa. Electronic circuits considered a
part of audio electronics may also be designed to achieve certain signal processing operations, in
order to make particular alterations to the signal while it is in the electrical form. [1] Additionally, audio
signals can be created synthetically through the generation of electric signals from electronic
devices. Audio Electronics were traditionally designed with analog electric circuit techniques until
advances in digital technologies were developed. Moreover, digital signals are able to be
manipulated by computer software much the same way audio electronic devices would, due to its
compatible digital nature. Both analog and digital design formats are still used today, and the use of
one or the other largely depends on the application.

Audio equipment refers to devices that reproduce, record, or process sound. This
includes microphones, radio receivers, AV receivers, CD players, tape recorders, amplifiers, mixing
consoles, effects units, and loudspeakers.[1]
Audio equipment is widely used in many different scenarios, such as concerts, bars, meeting
rooms and the home where there is a need to reproduce, record and enhance sound volume.

Sound system
Sound system may refer to:

Technology
 Sound reinforcement system, a system for amplifying audio for an audience
 High fidelity, a sound system intended for accurate reproduction of music in the
home
 Public address system, an institutional speech-reinforcement or public safety
announcement system
 Shelf stereo, a compact sound system for personal use

Arts and entertainment

 LCD Soundsystem, an American rock band
 Sound system (DJ), a group of disc jockeys performing together
 Sound system (Jamaican), a group of disc jockeys, engineers and MCs playing ska,
rocksteady or reggae music
 Sound System (album), a compilation album by The Clash
 Sound-System (album), a 1984 album by Herbie Hancock
 Soundsystem (311 album), a 1999 album by the American rock band 311
 Sound System Records, an Australian record label
 "Sound System", a song by Operation Ivy (band)


Sound reinforcement system


 Large outdoor pop music concerts use complex and powerful sound reinforcement systems.

 A sound reinforcement system is the combination of microphones, signal

processors, amplifiers, and loudspeakers in enclosures all controlled by a mixing
console that makes live or pre-recorded sounds louder and may also distribute
those sounds to a larger or more distant audience.[1][2] In many situations, a sound
reinforcement system is also used to enhance or alter the sound of the sources
on the stage, typically by using electronic effects, such as reverb, as opposed to
simply amplifying the sources unaltered.
 A sound reinforcement system for a rock concert in a stadium may be very
complex, including hundreds of microphones, complex live sound mixing and
signal processing systems, tens of thousands of watts of amplifier power, and
multiple loudspeaker arrays, all overseen by a team of audio engineers and
technicians. On the other hand, a sound reinforcement system can be as simple
as a small public address (PA) system, consisting of, for example, a single
microphone connected to a 100 watt amplified loudspeaker for a singer-guitarist
playing in a small coffeehouse. In both cases, these systems reinforce sound to
make it louder or distribute it to a wider audience.[3]
 Some audio engineers and others in the professional audio industry disagree
over whether these audio systems should be called sound reinforcement (SR)
systems or PA systems. Distinguishing between the two terms by technology and
 capability is common, while others distinguish by intended use (e.g., SR systems
are for live event support and PA systems are for reproduction of speech and
recorded music in buildings and institutions). In some regions or markets, the
distinction between the two terms is important, though the terms are considered
interchangeable in many professional circles

High fidelity
 High fidelity (often shortened to hi-fi or hifi) is a term used by listeners, audiophiles,
and home audio enthusiasts to refer to high-quality reproduction of sound.[1] This is in
contrast to the lower quality sound produced by inexpensive audio equipment, AM radio, or
the inferior quality of sound reproduction that can be heard in recordings made until the late
1940s.
 Ideally, high-fidelity equipment has inaudible noise and distortion, and a flat (neutral,
uncolored) frequency response within the human hearing range.

Types of audio equipment

Audio equipment refers to devices that reproduce, record, or process sound. This
includes microphones, radio receivers, AV receivers, CD players, tape recorders,
amplifiers, mixing consoles, effects units, and loudspeakers.

What are the 6 parts of a sound system?

In brief, a basic sound system consists of these components:

 Microphone (wired and/or wireless)
 Mixer.
 Power amplifier.
 Loudspeaker.
 All the necessary cabling.

Whats is a microphone?
A microphone is a device that translates sound vibrations in the air into electronic
signals or scribes them to a recording medium. Microphones enable many types of
audio recording devices for purposes including communications of many kinds, as well
as music and speech recording.

Microphone
.

Shure Brothers microphone, model 55s, Multi-Impedance "Small Unidyne" Dynamic from 1951

A Sennheiser dynamic microphone

A microphone, colloquially called a mic or mike (/maɪk/), is a device – a transducer –

that converts sound into an electrical signal. Microphones are used in many applications
such as telephones, hearing aids, public address systems for concert halls and public
events, motion picture production, live and recorded audio engineering, sound
recording, two-way radios, megaphones, radio and television broadcasting. They are
also used in computers for recording voice, speech recognition, VoIP, and for non-
acoustic purposes such as ultrasonic sensors or knock sensors.
Several types of microphone are used today, which employ different methods to convert
the air pressure variations of a sound wave to an electrical signal. The most common
are the dynamic microphone, which uses a coil of wire suspended in a magnetic field;
the condenser microphone, which uses the vibrating diaphragm as a capacitor plate;
and the contact microphone, which uses a crystal of piezoelectric material. Microphones
typically need to be connected to a preamplifier before the signal can be recorded or
reproduced.

Dynamic

Patti Smith singing into a Shure SM58 (dynamic cardioid type) microphone

The dynamic microphone (also known as the moving-coil microphone) works

via electromagnetic induction. They are robust, relatively inexpensive and resistant to moisture. This,
coupled with their potentially high gain before feedback, makes them ideal for on-stage use.
Dynamic microphones use the same dynamic principle as in a loudspeaker, only reversed. A small
movable induction coil, positioned in the magnetic field of a permanent magnet, is attached to
the diaphragm. When sound enters through the windscreen of the microphone, the sound wave
moves the diaphragm. When the diaphragm vibrates, the coil moves in the magnetic field, producing
a varying current in the coil through electromagnetic induction. A single dynamic membrane does not
respond linearly to all audio frequencies. For this reason, some microphones utilize multiple
membranes for the different parts of the audio spectrum and then combine the resulting signals.
Combining the multiple signals correctly is difficult; designs that do this are rare and tend to be
expensive. On the other hand, there are several designs that are more specifically aimed towards
isolated parts of the audio spectrum. The AKG D112, for example, is designed for bass response
rather than treble.[24] In audio engineering several kinds of microphones are often used at the same
time to get the best results.
Dynamic microphones are mostly used for live stage performances. They are
ideal choices for many musicians, professional broadcasts, radio stations and
live streaming.

Dynamic microphones are well known for their great durability and built quality.
They are moisture-resistant and available at a reasonable price.

What is a dynamic microphone?

Microphones are basically transducers that convert one form of energy into
another form.Simply, dynamic microphones convert sound waves into
electrical energy that is the voltage. Dynamic microphones work on the
principle of electromagnetic induction.

The diaphragm is attached to the voice coil and that voice coil is attached to the
magnet further to create the magnetic field.

Diaphragm: Usually made up of lightweight material like aluminum alloy

Voice Coil: A coil of wire commonly made of copper or aluminum

Magnet: To create a magnetic field with the help of a coil of wire

Transformer: Normally step-up transformer to get the high output impedance

XLR Assembly: For further XLR connection

How does a dynamic microphone work?

When we are using the dynamic mic, the sound hits the diaphragm and the
diaphragm starts vibrating forward and backward direction.

Simultaneously attached voice coil starts moving with respect to the stationary
magnet & which generates the magnetic field in the gap between the coil and the
magnet.

The magnetic field gives the output electrical signal that is a voltage.

Now we got voltage but that’s unbalanced and weak. The Step-up transformer
converts the unbalanced signal to the balanced one and steps up the voltage.

Further, we can use it through the XLR assembly which gives us an XLR
connection.

As our sound frequency changes, the pressure on the diaphragm changes.

Simultaneously output signal gets varied.

That’s the quick answer to how does a dynamic microphone work.

Types of dynamic microphones

There are two types of dynamic microphones. We have discussed earlier the first
one which is the moving coil microphone.

The second one is the ribbon microphone.

The structure of this microphone is as same as the previous one except that, a
metal ribbon is suspended in the magnetic field instead of the voice coil. That
ribbon is connected to the mic’s output.

Directionality of dynamic microphones

Dynamic microphones are sensitive to directions from where the sound is
coming. There are three types according to the sound directions.

Omnidirectional Microphones

Omnidirectional microphones are sensitive to every direction. They pick up the

sound from all sides.

Unidirectional Microphones

Unidirectional microphones are sensitive to only one direction.

Bidirectional Microphones

Bidirectional microphones are sensitive to two opposite directions. They pick the
audio signal equally from both opposite directions. Ribbon microphones are an
example of bidirectional microphones.

Advantages of dynamic microphones

 Dynamic microphones have great metallic construction. They are sturdy and
durable. Almost every dynamic microphone doesn’t get affected in the case of
falling from hands.

 Dynamic microphones are experts in auto rejecting background

noise. Cardioid polar pattern picks up the more of sound from the front of the
microphone less from the sides and rear of the microphone.

 Dynamic microphones don’t need any external power supply.

 Dynamic microphones give you deep audio. They can handle high pressure
levels of sound.

 Dynamic microphones are inexpensive. You can get them at cheap rates.
 Disadvantages of dynamic microphones
 Dynamic microphones are not suitable for high-frequency applications.

Frequently Asked Questions

What is a dynamic microphone used for?
Most of the time, a dynamic microphone is used for live stage performances and
streaming. As it’s great at handling ambient noise, it’s also used for recording
professional broadcasts.

Do dynamic microphones require power?

No. Most of the dynamic microphones do not require any external power supply.
But some of the dynamic microphones are an exception to that.

How do I connect a dynamic microphone to

my laptop?
Most of the dynamic microphones have XLR output. You’ll need an XLR to USB
cable for connection. Plug the USB end of the cable into the USB port. Connect
the XLR end to the microphone.

In the case of a USB output of the mic, you’ll need a USB to USB cable.

Can I use a dynamic microphone for

recording?
Yes. A dynamic microphone is a great choice for recording. You can use them.
 Conclusion
We discussed what is a dynamic microphone & how does a dynamic microphone
work.

Dynamic microphones are low sensitive as compared to condenser microphones.

But condenser microphones easily capture the noise.

If you don’t have a soundproof room, a dynamic microphone can be the right
choice for you.

Carbon microphone basics

The basic concept behind the carbon microphone is the fact that when carbon granules are
compressed their resistance decreases. This occurs because the granules come into better contact
with each other when they are pushed together by the higher pressure.

The carbon microphone comprises carbon granules that are contained within a small contained that
is covered with a thin metal diaphragm. A battery is also required to cause a current to flow through
the microphone.

When sound waves strike the carbon microphone diaphragm it vibrates, exerting a varying pressure
onto the carbon. These varying pressure levels are translated into varying levels of resistance, which
in turn vary the current passing through the microphone.

Construction of a carbon microphone

The varying current can be passed through a transformer or a capacitor to enable it to be used
within a telephone, or by some form of amplifier.

The frequency response of the carbon microphone, however, is limited to a narrow range, and the
device produces significant electrical noise. Often the microphone would produce a form of crackling
noise which could be eliminated by shaking it or giving it a small sharp knock. This would shake the
carbon granules and enable them to produce a more steady current.

Carbon microphone applications

Carbon microphones were an ideal choice of microphone in the early days of the telephone. They
were widely used in telephone applications because they gave a high output which meant no
amplification was used.

Carbon microphones were used in

telephones like this vintage British GPO 300 series telephone
As radio started to be used, the carbon microphone was initially used there as well – for
broadcasting as well as communications purposes. However their use in broadcast applications
soon came to end because of the drawbacks of noise and poor frequency response. Other types of
microphone started to become available and their use was preferred because of the better fidelity
that was available. The use of the carbon microphone persisted for many years for communications
purposes as they gave a high output and they were robust. The poor frequency response was not an
issue.

The carbon microphone was used for telephones up until the 1970s and 1980s, but even there it
became possible to use other types of microphone more conveniently. Also the crackle and noise of
the carbon microphone had always been an issue and when other types of microphone became
available at a low cost they started t be used, despite the requirement for additional electronics
needed.

Carbon microphones are now only used in a very few applications – typically only specialist
applications. They are able to withstand high voltage spikes and this property lends itself to use in a
small number of applications.

Carbon microphone advantages &

disadvantages
As with any form of microphone there are advantages and disadvantages.

Carbon microphone advantages

 High output
 Simple principle & construction
 Cheap and simple to manufacture
Carbon microphone disadvantages
 Very noisy - high background noise and on occasions it would crackle
 Poor frequency response
 Requires battery or other supply for operation.

The carbon microphone has a number of advantages, but today the disadvantages normally
outweigh the positives and as a result they are rarely used..

Condenser
Inside the Oktava 319 condenser microphone

The condenser microphone, invented at Western Electric in 1916 by E. C.

Wente,is also called a capacitor microphone or electrostatic microphone—
capacitors were historically called condensers. Here, the diaphragm acts as one
plate of a capacitor, and the vibrations produce changes in the distance
between the plates. There are two types, depending on the method of extracting
the audio signal from the transducer: DC-biased microphones, and radio
frequency (RF) or high frequency (HF) condenser microphones. With a DC-
biased microphone, the plates are biased with a fixed charge (Q).
The voltage maintained across the capacitor plates changes with the vibrations
in the air, according to the capacitance equation (C = Q⁄V), where Q = charge
in coulombs, C = capacitance in farads and V = potential difference in volts.
The capacitance of the plates is inversely proportional to the distance between
them for a parallel-plate capacitor. The assembly of fixed and movable plates is
called an "element" or "capsule".
A nearly constant charge is maintained on the capacitor. As the capacitance
changes, the charge across the capacitor does change very slightly, but at
audible frequencies it is sensibly constant. The capacitance of the capsule
(around 5 to 100 pF) and the value of the bias resistor (100 MΩ to tens of GΩ)
form a filter that is high-pass for the audio signal, and low-pass for the bias
voltage. Note that the time constant of an RC circuit equals the product of the
resistance and capacitance.
Within the time-frame of the capacitance change (as much as 50 ms at 20 Hz
audio signal), the charge is practically constant and the voltage across the
capacitor changes instantaneously to reflect the change in capacitance. The
voltage across the capacitor varies above and below the bias voltage. The
voltage difference between the bias and the capacitor is seen across the series
resistor. The voltage across the resistor is amplified for performance or
recording. In most cases, the electronics in the microphone itself contribute no
voltage gain as the voltage differential is quite significant, up to several volts for
high sound levels. Since this is a very high impedance circuit, only current gain
is usually needed, with the voltage remaining constant.
 AKG C451B small-diaphragm condenser microphone

RF condenser microphones use a comparatively low RF voltage, generated by

a low-noise oscillator. The signal from the oscillator may either be amplitude
modulated by the capacitance changes produced by the sound waves moving
the capsule diaphragm, or the capsule may be part of a resonant circuit that
modulates the frequency of the oscillator signal. Demodulation yields a low-
noise audio frequency signal with a very low source impedance. The absence of
a high bias voltage permits the use of a diaphragm with looser tension, which
may be used to achieve wider frequency response due to higher compliance.
The RF biasing process results in a lower electrical impedance capsule, a
useful by-product of which is that RF condenser microphones can be operated
in damp weather conditions that could create problems in DC-biased
microphones with contaminated insulating surfaces. The Sennheiser "MKH"
series of microphones use the RF biasing technique. A covert, remotely
energised application of the same physical principle was devised by Soviet
Russian inventor Leon Theremin and used to bug the US Ambassador's
Residence in Moscow between 1945 and 1952.
Condenser microphones span the range from telephone transmitters through
inexpensive karaoke microphones to high-fidelity recording microphones. They
generally produce a high-quality audio signal and are now the popular choice
in laboratory and recording studio applications. The inherent suitability of this
technology is due to the very small mass that must be moved by the incident
sound wave, unlike other microphone types that require the sound wave to do
more work. They require a power source, provided either via microphone inputs
on equipment as phantom power or from a small battery. Power is necessary
for establishing the capacitor plate voltage and is also needed to power the
microphone electronics (impedance conversion in the case of electret and DC-
polarized microphones, demodulation or detection in the case of RF/HF
microphones). Condenser microphones are also available with two diaphragms
that can be electrically connected to provide a range of polar patterns (see
below), such as cardioid, omnidirectional, and figure-eight. It is also possible to
vary the pattern continuously with some microphones, for example,
the Røde NT2000 or CAD M179.
A valve microphone is a condenser microphone that uses a vacuum
tube (valve) amplifier. They remain popular with enthusiasts of tube sound.
1) Sound Waves 2) Diaphragm 3) Back Plate 4) Battery 5) Resistor 6) Audio
Signal (Image: Wikicommons 3.0)

Unlike dynamic microphones, condenser microphones are capable of capturing

those much quieter sounds with a high degree of accuracy.

A condenser microphone also contains a diaphragm, which is usually made of

very thin metal and another piece of metal called a backplate. Electricity is
applied to both of these creating a static charge between them.

Once a soundwave hits the diaphragm it vibrates and produces a small

electrical current.

Used for: quieter more complex sounds with a greater range of frequencies

Pros: sensitive, accurate

Cons: more expensive, more delicate, don’t deal well with very loud sounds
1) incoming sound 2) diaphragm 3) coil 4) permanent magnet 5) resulting signal (Image:
wiki commons CC3.0)

Used for: loud sounds, live instruments/ amps, drums

Pros: cheap, durable, doesn’t need a power source

Cons: not very sensitive to quiet or high-frequency sounds

A dynamic microphone is the oldest type of microphone and is thus the most primitive in
terms of design.

In very simplified terms, the sound in a dynamic microphone is created when a sound
wave hits a diaphragm (a device usually made of plastic or polyester film used to sense
a sound signal) causing it to move.

The diaphragm is attached to a metal coil which is suspended between two magnets.
When the diaphragm moves the coil also moves up and down producing a small AC
current, mimicking that of the sound wave.

To try and make this easier to understand, imagine the sound wave being like a wave
on the water that you create by splashing, and then imagine the metal coil as a cork
bobbing up and down on the surface as each wave passes it…

Dynamic microphones are capable of withstanding high sound pressure levels. This
makes them ideal for recording loud sounds or for use in a live setting. They are also
extremely reasonably priced due to their fairly rudimentary design and they can
withstand a lot of wear and tear. This is one of the reasons they are the most frequently
used microphone for live performances.
This durability becomes a limitation of dynamic microphones in some situations.

The coil has a certain weight to it and therefore if you make a quiet sound or perhaps a
sound of particularly high or low frequency, the coil will not vibrate sufficiently to
produce an accurate representation of the sound.

So in a studio, where you aren’t worried about sounds being particularly loud, and
where you want to record the intricacies of your vocals. A dynamic microphone may not
be the best fit.

And that is when you may need…….a condenser microphone

Electret Microphone

An electret microphone is a type of condenser microphone invented by Gerhard

Sessler and Jim West at Bell laboratories in 1962. The externally applied
charge used for a conventional condenser microphone is replaced by a
permanent charge in an electret material. An electret is a ferroelectric material
that has been permanently electrically charged or polarized. The name comes
from electrostatic and magnet; a static charge is embedded in an electret by the
alignment of the static charges in the material, much the way a permanent
magnet is made by aligning the magnetic domains in a piece of iron.
Due to their good performance and ease of manufacture, hence low cost, the
vast majority of microphones made today are electret microphones; a
semiconductor manufacturer estimates annual production at over one billion
units. They are used in many applications, from high-quality recording
and lavalier (lapel mic) use to built-in microphones in small sound
recording devices and telephones. Prior to the proliferation of MEMS
microphones, nearly all cell-phone, computer, PDA and headset microphones
were electret types.
Unlike other capacitor microphones, they require no polarizing voltage, but
often contain an integrated preamplifier that does require power (often
incorrectly called polarizing power or bias). This preamplifier is
frequently phantom powered in sound reinforcement and studio applications.
Monophonic microphones designed for personal computers (PCs), sometimes
called multimedia microphones, use a 3.5 mm plug as usually used, without
power, for stereo; the ring, instead of carrying the signal for a second channel,
carries power via a resistor from (normally) a 5 V supply in the computer.
Stereophonic microphones use the same connector; there is no obvious way to
determine which standard is used by equipment and microphones.
Though electret microphones were once considered low quality, the best ones
can now rival traditional condenser microphones in every respect and can even
offer the long-term stability and ultra-flat response needed for a measurement
microphone. Only the best electret microphones rival good DC-polarized units
in terms of noise level and quality; electret microphones lend themselves to
inexpensive mass-production, while inherently expensive non-electret
condenser microphones are made to higher quality.

How Electret Microphone Works?

An electret microphone is a type of electrostatic capacitor-based electronics
component, which can be receiving sound waves and transforming it into tiny
electrical pulses. An electret microphone mainly consists of an Electret
Capacitor, a JFET, and a resistor. The sound waves mainly receive by the
Electret Capacitor. Electret Capacitor is a variable capacitor when the sound
wave comes on it, then the capacitance will vary in the capacitor. It has two
terminals one is Negative and another one is Positive. It is able to receive
sound vibrations from across all angles.

Electret Microphone internal circuit structure

This device has a microphone and pre-amplifier built into a small enclosure. It
is important to connect it the right way round because the built-in preamp
might be destroyed otherwise. The preamp uses a field effect transistor (FET).
This microphone uses two metal plates as a capacitor. One of the plates moves
or vibrates in step with incoming sound waves. This causes the capacitance to
change.
A Mathematical Explanation
Q = CV
Q = charge (Coulombs)
C = capacitance (Farads)
V = voltage (Volts)
Since Q stays constant (it does not have time to change) and since C changes,
it follows that V will change. Thus incoming sound waves are converted into a
proportional alternating voltage. This voltage is amplified by the built-in FET.
This is a delicate, easily damaged component.

Electret Microphone Circuit

The resistor to the positive supply is not well described in the data sheets.
(Supply_Voltage x 3) kΩ seems to work so in the circuit below, 33kΩ to 47kΩ
should be used.
Pre-Amplifier Circuit

This can be used to amplify a microphone signal before passing the output to
an audio power amplifier or perhaps to the sound-card in a PC.

Piezoelectric

Vintage Astatic crystal microphone

A crystal microphone or piezo microphone uses the phenomenon

of piezoelectricity—the ability of some materials to produce a voltage when
subjected to pressure—to convert vibrations into an electrical signal. An
example of this is potassium sodium tartrate, which is a piezoelectric crystal
that works as a transducer, both as a microphone and as a slimline
loudspeaker component. Crystal microphones were once commonly supplied
with vacuum tube (valve) equipment, such as domestic tape recorders. Their
high output impedance matched the high input impedance (typically about
10 megohms) of the vacuum tube input stage well. They were difficult to match
to early transistor equipment and were quickly supplanted by dynamic
microphones for a time, and later small electret condenser devices. The high
impedance of the crystal microphone made it very susceptible to handling
noise, both from the microphone itself and from the connecting cable.
Piezoelectric transducers are often used as contact microphones to amplify
sound from acoustic musical instruments, to sense drum hits, for triggering
electronic samples, and to record sound in challenging environments, such as
underwater under high pressure. Saddle-mounted pickups on acoustic
guitars are generally piezoelectric devices that contact the strings passing over
the saddle. This type of microphone is different from magnetic coil
pickups commonly visible on typical electric guitars, which use magnetic
induction, rather than mechanical coupling, to pick up vibration.

How Piezoelectricity Works

We have specific materials that are suited for piezoelectricity

applications, but how exactly does the process work? With the
Piezoelectric Effect. The most unique trait of this effect is that it works
two ways. You can apply mechanical energy or electrical energy to the
same piezoelectric material and get an opposite result.
Applying mechanical energy to a crystal is called a direct piezoelectric
effect and works like this:

1. A piezoelectric crystal is placed between two metal plates. At this

point the material is in perfect balance and does not conduct an
electric current.
2. Mechanical pressure is then applied to the material by the metal
plates, which forces the electric charges within the crystal out of
balance. Excess negative and positive charges appear on opposite
sides of the crystal face.
3. The metal plate collects these charges, which can be used to
produce a voltage and send an electrical current through a circuit.

That’s it, a simple application of mechanical pressure, the squeezing of a

crystal and suddenly you have an electric current. You can also do the
opposite, applying an electrical signal to a material as an inverse
piezoelectric effect. It works like this:

1. In the same situation as the example above, we have a piezoelectric

crystal placed between two metal plates. The crystal’s structure is in
perfect balance.
2. Electrical energy is then applied to the crystal, which shrinks and
expands the crystal’s structure.
 3. As the crystal’s structure expands and contracts, it converts the received
electrical energy and releases mechanical energy in the form of a sound
wave.

Ribbon microphone
 (left) RCA "44-BX" ribbon microphone from 1940. (right) RCA 44-type With
the cover off. The magnet is visible at center, and the narrow aluminum
ribbon is suspended between the triangular pole pieces (top).

A ribbon microphone, also known as a ribbon velocity microphone, is a type

of microphone that uses a thin aluminum, duraluminum or nanofilm of
electrically conductive ribbon placed between the poles of a magnet to produce
a voltage by electromagnetic induction. Ribbon microphones are typically
bidirectional, meaning that they pick up sounds equally well from either side of
the microphone.

Principle of operation:
The sensitivity pattern of a bidirectional microphone (red dot) viewed from
above.

In a moving-coil microphone, the diaphragm is attached to a light movable coil

that generates a voltage as it moves back and forth between the poles of
a permanent magnet. In ribbon microphones, a light metal ribbon (usually
corrugated) is suspended between the poles of a magnet. As the ribbon
vibrates, a voltage is induced at right angles to both the ribbon velocity
and magnetic field direction and is picked off by contacts at the ends of the
ribbon. Ribbon microphones are also called "velocity microphones" because the
induced voltage is proportional to the velocity of the ribbon and thus of the air
particles in the sound wave, unlike in some other microphones where the
voltage is proportional to the displacement of the diaphragm and the air.
One important advantage that the ribbon microphone had when it was
introduced is that its very lightweight ribbon, which is under very little tension,
has a resonant frequency lower than 20 Hz; in contrast to the typical resonant
frequency of the diaphragms in contemporary high quality microphones which
used other technology. The typical resonant frequency of those microphones is
within the range of human hearing. So even the very early commercially
available ribbon microphones had excellent frequency response throughout the
nominal range of human hearing (20 Hz to 20 kHz for a young adult).
The voltage output of older ribbon microphones is typically quite low compared
to a dynamic moving coil microphone, and a step-up transformer is used to
increase the voltage output and increase the output impedance. Modern ribbon
microphones do not suffer from this problem due to improved magnets and
more efficient transformers and have output levels that can exceed typical
stage dynamic microphones.
 Principle of operation

Ribbon microphones were once delicate and expensive, but modern materials
make certain present-day ribbon microphones very durable, and so they may
be used for loud rock music and stage work. They are prized for their ability to
capture high-frequency detail, comparing very favorably with condenser
microphones, which can often sound subjectively "aggressive" or "brittle" in the
high end of the frequency spectrum. Due to their bidirectional pick-up pattern,
ribbon microphones may be used in pairs to produce the Blumlein
Pair recording array. In addition to the standard bidirectional pick-up
pattern, ribbon microphones can also be configured to
have cardioid, hypercardioid,and variable pattern.
As many mixers are equipped with phantom power in order to enable the use
of condenser microphones, care should be taken when using condenser and
ribbon microphones at the same time. If the ribbon microphone is improperly
wired, which is not unheard of with older microphones, this capability can
damage some ribbon elements; however, improvements in designs and
materials have made those concerns largely inconsequential in modern ribbon
microphones.

Fiber-optic
The Optoacoustics 1140 fiber-optic microphone

A fiber-optic microphone converts acoustic waves into electrical signals by sensing changes in light
intensity, instead of sensing changes in capacitance or magnetic fields as with conventional
microphones.
During operation, light from a laser source travels through an optical fiber to illuminate the surface of
a reflective diaphragm. Sound vibrations of the diaphragm modulate the intensity of light reflecting
off the diaphragm in a specific direction. The modulated light is then transmitted over a second
optical fiber to a photodetector, which transforms the intensity-modulated light into analog or digital
audio for transmission or recording. Fiber-optic microphones possess high dynamic and frequency
range, similar to the best high fidelity conventional microphones.
Fiber-optic microphones do not react to or influence any electrical, magnetic, electrostatic or
radioactive fields (this is called EMI/RFI immunity). The fiber-optic microphone design is therefore
ideal for use in areas where conventional microphones are ineffective or dangerous, such as
inside industrial turbines or in magnetic resonance imaging (MRI) equipment environments.
Fiber-optic microphones are robust, resistant to environmental changes in heat and moisture, and
can be produced for any directionality or impedance matching. The distance between the
microphone's light source and its photodetector may be up to several kilometers without need for
any preamplifier or another electrical device, making fiber-optic microphones suitable for industrial
and surveillance acoustic monitoring.
Fiber-optic microphones are used in very specific application areas such as
for infrasound monitoring and noise-canceling. They have proven especially useful in medical
applications, such as allowing radiologists, staff and patients within the powerful and noisy magnetic
field to converse normally, inside the MRI suites as well as in remote control rooms. Other uses
include industrial equipment monitoring and audio calibration and measurement, high-fidelity
recording and law enforcement.
FIBER OPTIC CONNECTORS
A New Kind Of Microphone Fiber Optic Cable

Usage

People use fiber optic microphones in very specific application

areas such as infrasound monitoring and noise-canceling. They
have proven especially useful in medical applications. They
allow radiologists, staff, and patients within the powerful and
noisy magnetic field to converse normally, inside the MRI
suites as well as in remote control rooms. Other uses include
industrial equipment monitoring and audio calibration and
measurement, high-fidelity recording and law enforcement.

In the medical field, for example, the optical microphone is ideally suited for
use in magnetic resonance imaging (MRI) in order to maintain contact with the
patient during MRI scans or to provide active noise cancellation. ... The
diaphragm reflects part of the light into a receiver fiber optic cable.
Laser microphone
A laser microphone is a surveillance device that uses a laser beam to
detect sound vibrations in a distant object. It can be used to eavesdrop with minimal
chance of exposure.
The object is typically inside a room where a conversation is taking place and can be
anything that can vibrate (for example, a picture on a wall) in response to the pressure
waves created by noises present in the room. The object preferably should have a
smooth surface for the beam to be reflected accurately. The laser beam is directed into
the room through a window, reflects off the object, and returns to a receiver that
converts the beam to an audio signal. The beam may also be bounced off the window
itself. The minute differences in the distance traveled by the light as it reflects from the
vibrating object are detected interferometrically. The interferometer converts the
variations to intensity variations, and electronics are used to convert these variations to
signals that can be converted back to sound.
The Laser Microphone
The human voice can generate sound waves in the range of 300 Hz to 3400 Hz [8]. These sound
waves vibrate nearby objects, making it possible for an analog electronic device to convert these
vibrations into an audio signal[1]. One way to accomplish this conversion from movement to audio is
to use a "laser microphone", which reflects a laser off the vibrating object and uses a receiver to
capture the laser's reflection. The reflection of the laser gets deflected as vibrations shift the surface
of the vibrating object. Therefore, if a receiver takes in the oscillating laser signal from a fixed
location, the receiver will detect the laser deflections caused by the vibrations that were originally
produced from an audio signal. The receiver can then filter and amplify this signal, and output it as
audio. Through this process the laser microphone effectively reproduces the audio that induced the
object's vibrations. The laser microphone is able to reproduce audio detected from a vibrating
surface with relatively high accuracy: less than 8% distortion. As an additional feature, the laser
microphone is also able to transmit audio via amplitude-modulated laser signal, capture the laser
signal, and output the audio. Thus by using a laser based system that captures oscillations in the
position of the laser, the laser microphone is able to accurately reproduce both the audio that
induced an object's vibrations and audio transmitted via laser communication.
MEMS
Main article: Microelectromechanical systems
The MEMS (microelectromechanical systems) microphone is also called a microphone chip or
silicon microphone. A pressure-sensitive diaphragm is etched directly into a silicon wafer by MEMS
processing techniques and is usually accompanied with an integrated preamplifier. Most MEMS
microphones are variants of the condenser microphone design. Digital MEMS microphones have
built-in analog-to-digital converter (ADC) circuits on the same CMOS chip making the chip a digital
microphone and so more readily integrated with modern digital products. Major manufacturers
producing MEMS silicon microphones are Wolfson Microelectronics (WM7xxx) now Cirrus
Logic, InvenSense (product line sold by Analog Devices ), Akustica (AKU200x), Infineon (SMM310
product), Knowles Electronics, Memstech (MSMx), NXP Semiconductors (division bought by
Knowles ), Sonion MEMS, Vesper, AAC Acoustic Technologies, and Omron.
More recently, since the 2010s, there has been increased interest and research into making
piezoelectric MEMS microphones which are a significant architectural and material change from
existing condenser style MEMS designs.

MEMS Microphone Basics

The MEMS diaphragm forms a capacitor and sound pressure waves cause movement
of the diaphragm. MEMS microphones typically contain a second semiconductor die
which functions as an audio preamplifier, converting the changing capacitance of the
MEMS to an electrical signal.
USES OF MEMS MICROPHONE:

MEMS microphones are ideal for wireless applications such as mobile phones, audio
monitoring WSNs, and sound source localization WSNs. These applications typically
require low power consumption, miniature sizes, and relatively high data quality.

. Wireless or Cordless Microphone • The other audio equipment is

connected to the receiver unit by cable. • Wireless microphones are
widely used in the entertainment industry, television broadcasting,
and public speaking to allow public speakers, interviewers,
performers, and entertainers to move about freely while using a
microphone to amplify their voices. • These are Hand held and
collar type as shown in figure
.
. Advantages • Greater freedom of movement for the artist or
speaker. • Avoidance of cabling stressing problems common with
wired microphones. • Reduction of cable "trip hazards" in the
performance space
.
. Disadvantages • Some wireless systems have a shorter range, while
more expensive models can exceed that distance. • Possible
interference with or, more often, from other radio equipment or
other radio microphones. • Operation time is limited relative to
battery life.
 . Loudspeakers
.
. Loudspeakers • Definition • A loudspeaker (or "speaker") is an electro‐
acoustic transducer that produces sound in response to an electrical
audio signal input. • Loudspeakers may be divided into two main groups: ‐
cone type and horn type
.
. Description of cone and horn speakers • 1) Cone type ‐ i.e., direct radiator,
where cone or diaphragm is directly coupled to air. • 2) Horn‐type‐ i.e.,
indirect radiator, where the diaphragm is coupled to the air by means of
horn. • The horn increases the acoustical loading on the diaphragm and
thereby increases the efficiency. • It may be described as a device, which
transforms acoustical energy at high pressure and low velocity to
acoustical energy at low pressure and high velocity.
.
. 1) Dynamic Loudspeaker: • The most common type of driver, commonly
called a dynamic loudspeaker. • It has a light weight diaphragm, or cone,
connected to a rigid basket, or frame, via a flexible suspension, commonly
called a spider, that constrains a coil of fine tensile wire to move axially
through a cylindrical magnetic gap.
.
. Dynamic Loudspeaker • When an electrical signal is applied to the voice
coil, a magnetic field is created by the electric current in the voice coil,
making it a variable electromagnet. • The coil and the driver's magnetic
system interact, generating a mechanical force that causes the coil (and
thus, the attached cone) to move back and forth, thereby reproducing
sound under the control of the applied electrical signal coming from the
amplifier.
. Dynamic Loudspeaker Cross section & Construction of Dynamic
Loudspeaker
. Dynamic Loudspeaker

Cutaway view of a dynamic loudspeaker for the bass register.

1. Magnet
 2. Voicecoil
3. Suspension
4. Diaphragm

Cutaway view of a dynamic midrange speaker.

1. Magnet
2. Cooler (sometimes present)
3. Voicecoil
4. Suspension
5. Diaphragm

Cutaway view of a dynamic tweeter with acoustic lens and a dome-shaped membrane.

1. Magnet
2. Voicecoil
3. Diaphragm
4. Suspension
 A four-way, high fidelity loudspeaker system. Each of the four drivers outputs a different frequency range; the
fifth aperture at the bottom is a bass reflex port.

Subwoofer
Main article: Subwoofer
A subwoofer is a woofer driver used only for the lowest-pitched part of the
audio spectrum: typically below 200 Hz for consumer systems,below 100 Hz for
professional live sound, and below 80 Hz in THX-approved systems. Because
the intended range of frequencies is limited, subwoofer system design is
usually simpler in many respects than for conventional loudspeakers, often
consisting of a single driver enclosed in a suitable box or enclosure. Since
sound in this frequency range can easily bend around corners by diffraction,
the speaker aperture does not have to face the audience, and subwoofers can
be mounted in the bottom of the enclosure, facing the floor. This is eased by
the limitations of human hearing at low frequencies; such sounds cannot be
located in space, due to their large wavelengths compared to higher frequencies
which produce differential effects in the ears due to shadowing by the head,
and diffraction around it, both of which we rely upon for localization clues.

Woofer
Main article: Woofer
A woofer is a driver that reproduces low frequencies. The driver works with the
characteristics of the enclosure to produce suitable low frequencies
(see speaker enclosure for some of the design choices available). Indeed, both
are so closely connected that they must be considered together in use. Only at
design time do the separate properties of enclosure and woofer matter
individually. Some loudspeaker systems use a woofer for the lowest
frequencies, sometimes well enough that a subwoofer is not needed.
Additionally, some loudspeakers use the woofer to handle middle frequencies,
eliminating the mid-range driver. This can be accomplished with the selection
of a tweeter that can work low enough that, combined with a woofer that
responds high enough, the two drivers add coherently in the middle
frequencies.
Mid-range driver
Main article: Mid-range speaker
A mid-range speaker is a loudspeaker driver that reproduces a band of
frequencies generally between 1–6 kHz, otherwise known as the 'mid'
frequencies (between the woofer and tweeter). Mid-range driver diaphragms can
be made of paper or composite materials, and can be direct radiation drivers
(rather like smaller woofers) or they can be compression drivers (rather like
some tweeter designs). If the mid-range driver is a direct radiator, it can be
mounted on the front baffle of a loudspeaker enclosure, or, if a compression
driver, mounted at the throat of a horn for added output level and control of
radiation pattern.

Tweeter

Exploded view of a dome tweeter

Main article: Tweeter

A tweeter is a high-frequency driver that reproduces the highest frequencies in a speaker
system. A major problem in tweeter design is achieving wide angular sound coverage (off-
axis response), since high frequency sound tends to leave the speaker in narrow beams.
Soft-dome tweeters are widely found in home stereo systems, and horn-loaded
compression drivers are common in professional sound reinforcement. Ribbon tweeters
have gained popularity in recent years, as the output power of some designs has been
increased to levels useful for professional sound reinforcement, and their output pattern is
wide in the horizontal plane, a pattern that has convenient applications in concert sound.

Crossover
Main article: Audio crossover
 A passive crossover

A bi-amplified system with an active crossover

Used in multi-driver speaker systems, the crossover is an assembly of filters

that separate the input signal into different frequency ranges (i.e. "bands"),
according to the requirements of each driver. Hence the drivers receive power
only at their operating frequency (the sound frequency range they were
designed for), thereby reducing distortion in the drivers and interference
between them. The ideal characteristics of a crossover may include perfect out-
of-band attenuation at the output of each filter, no amplitude variation
("ripple") within each passband, no phase delay between overlapping frequency
bands, to name just a few.
Crossovers can be passive or active. A passive crossover is an electronic circuit
that uses a combination of one or more resistors, inductors, or non-
polar capacitors. These components are combined to form a filter network and
are most often placed between the full frequency-range power amplifier and the
loudspeaker drivers to divide the amplifier's signal into the necessary frequency
bands before being delivered to the individual drivers. Passive crossover
circuits need no external power beyond the audio signal itself, but have some
disadvantages: they may require larger inductors and capacitors due to power
handling requirements (being driven by the amplifier), limited component
availability to optimize the crossover's characteristics at such power levels, etc.
Unlike active crossovers which include a built-in amplifier, passive crossovers
have an inherent attenuation within the passband, typically leading to a
reduction in damping factor before the voice coil. An active crossover is an
electronic filter circuit that divides the signal into individual frequency
bands before power amplification, thus requiring at least one power amplifier
for each bandpass. Passive filtering may also be used in this way before power
amplification, but it is an uncommon solution, being less flexible than active
filtering. Any technique that uses crossover filtering followed by amplification is
commonly known as bi-amping, tri-amping, quad-amping, and so on,
depending on the minimum number of amplifier channels.
Some loudspeaker designs use a combination of passive and active crossover
filtering, such as a passive crossover between the mid- and high-frequency
drivers and an active crossover between the low-frequency driver and the
combined mid- and high frequencies.
Passive crossovers are commonly installed inside speaker boxes and are by far
the most usual type of crossover for home and low-power use. In car audio
systems, passive crossovers may be in a separate box, necessary to
accommodate the size of the components used. Passive crossovers may be
simple for low-order filtering, or complex to allow steep slopes such as 18 or
24 dB per octave. Passive crossovers can also be designed to compensate for
undesired characteristics of driver, horn, or enclosure resonances, and can be
tricky to implement, due to component interaction. Passive crossovers, like the
driver units that they feed, have power handling limits, have insertion losses
(10% is often claimed), and change the load seen by the amplifier. The changes
are matters of concern for many in the hi-fi world.When high output levels are
required, active crossovers may be preferable. Active crossovers may be simple
circuits that emulate the response of a passive network, or may be more
complex, allowing extensive audio adjustments. Some active crossovers,
usually digital loudspeaker management systems, may include electronics and
controls for precise alignment of phase and time between frequency bands,
equalization, dynamic range compression and limiting control.
 Enclosures
Main article: Loudspeaker enclosure

An unusual three-way speaker system. The cabinet is narrow to raise the

frequency where a diffraction effect called the "baffle step" occurs.

Most loudspeaker systems consist of drivers mounted in an enclosure, or

cabinet. The role of the enclosure is to prevent sound waves emanating from
the back of a driver from interfering destructively with those from the front.
The sound waves emitted from the back are 180° out of phase with those
emitted forward, so without an enclosure they typically cause cancellations
which significantly degrade the level and quality of sound at low frequencies.
The simplest driver mount is a flat panel (i.e., baffle) with the drivers mounted
in holes in it. However, in this approach, sound frequencies with a wavelength
longer than the baffle dimensions are canceled out, because the antiphase
radiation from the rear of the cone interferes with the radiation from the front.
With an infinitely large panel, this interference could be entirely prevented. A
sufficiently large sealed box can approach this behavior.
Since panels of infinite dimensions are impossible, most enclosures function by
containing the rear radiation from the moving diaphragm. A sealed enclosure
prevents transmission of the sound emitted from the rear of the loudspeaker by
confining the sound in a rigid and airtight box. Techniques used to reduce
transmission of sound through the walls of the cabinet include thicker cabinet
walls, lossy wall material, internal bracing, curved cabinet walls—or more
rarely, visco-elastic materials (e.g., mineral-loaded bitumen) or
thin lead sheeting applied to the interior enclosure walls.
However, a rigid enclosure reflects sound internally, which can then be
transmitted back through the loudspeaker diaphragm—again resulting in
degradation of sound quality. This can be reduced by internal absorption using
absorptive materials (often called "damping"), such as glass wool, wool, or
synthetic fiber batting, within the enclosure. The internal shape of the
enclosure can also be designed to reduce this by reflecting sounds away from
the loudspeaker diaphragm, where they may then be absorbed.
Other enclosure types alter the rear sound radiation so it can add
constructively to the output from the front of the cone. Designs that do this
(including bass reflex, passive radiator, transmission line, etc.) are often used
to extend the effective low-frequency response and increase low-frequency
output of the driver.
To make the transition between drivers as seamless as possible, system
designers have attempted to time-align (or phase adjust) the drivers by moving
one or more driver mounting locations forward or back so that the acoustic
center of each driver is in the same vertical plane. This may also involve tilting
the face speaker back, providing a separate enclosure mounting for each
driver, or (less commonly) using electronic techniques to achieve the same
effect. These attempts have resulted in some unusual cabinet designs.
The speaker mounting scheme (including cabinets) can also cause diffraction,
resulting in peaks and dips in the frequency response. The problem is usually
greatest at higher frequencies, where wavelengths are similar to, or smaller
than, cabinet dimensions. The effect can be minimized by rounding the front
edges of the cabinet, curving the cabinet itself, using a smaller or narrower
enclosure, choosing a strategic driver arrangement, using absorptive material
around a driver, or some combination of these and other schemes.
.
. 2) Cabinet Loud Speaker • The cabinet improves the acoustic response
of the cone type speakers. • The basic design consists of an enclosure
with the loudspeaker unit set in the centre of a large box, which is
completely air tight except for a port and the loudspeaker hole in the
front panel. • The port is so proportioned to the interior volume of the
enclosure and to the loudspeaker characteristics that it functions
acoustically as a low frequency loudspeaker.
. Cabinet Loud Speaker • Thus, the low frequency response is increased,
and distortion generally experienced with a no ported enclosure, is
reduced. • The resonant frequency of a loudspeaker enclosure is damped
by completely lining the interior surfaces of the enclosure with a highly
absorbent material such as, rock wool. • The resonant frequency of the
panels may be damped to the use of diagonal braces and by filling
unused spaces with sand.
. Cabinet Loud Speaker
. 3) Line Source or Column Speaker
. Line Source or Column Speaker • Column Speakers use multiple
speaker cones create a slim line column offering excellent vertical sound
dispersion with a long 'throw', but limited horizontal coverage. • For this
reason, several column speakers can be mounted in a cluster and are
often used around pillars for sound reinforcement.
. Line Source or Column Speaker • On the axis of the system the sound
waves from all the units are in phase and will therefore reinforce each
other. • Off this axis the different path lengths from the units will tend to
cause cancellation. • However it will show that phase cancellation can
only occur if the wavelengths are comparable with or less than, the
length of loudspeaker column.
. 4) High Fidelity (Hi‐Fi) Speaker • These are used to reproduce the
generally audible frequency range of 50 Hz to 12 KHz (out of the entire
audio range of 20 Hz to 20 KHz). • The frequency response of ordinary
 speakers is irregular, with a number of resonant peaks and valleys, and
has a range of about 60 Hz to 8 KHz only.
. High Fidelity (Hi‐Fi) Speaker • By using a fairly large (30cm to 38 cm
diameter) and heavy cone, the low frequency response of speakers can be
extended downward to 45 or even 30 Hz but at the cost of high frequency
response. • It is difficult to design a single speaker to cover the entire
audio range. • One can use separate speakers for different audio ranges or
combine large and small speakers into a single unit, mounted in line or
coaxially.
. . * What are woofers • Woofer is designed to produce low frequency
sounds, typically from around 40 hertz up to about a kilohertz or higher. •
The most common design for a woofer is the electro dynamic driver, which
typically uses a stiff paper cone, driven by a voice coil which is
surrounded by a magnetic field.
. Woofer • The voice coil is attached by adhesives to the back of the
speaker cone. • The voice coil and magnet form a linear electric motor. •
When current flows through the voice coil, the coil moves in relation to the
frame according to Fleming's left hand rule, causing the coil to push or pull
on the driver cone in a piston‐like way. • The resulting motion of the cone
creates sound waves as it moves in and out.
. Woofer
. . * Tweeter
. Tweeter • A tweeter is a loudspeaker designed to produce high audio
frequencies, typically from around 2,000 Hz to 20,000 Hz (generally
considered to be the upper limit of human hearing). • Specialty tweeters
can deliver high frequencies up to 100 kHz.
. Tweeter • Tweeter in a two speaker system re‐produces frequencies from
1KHz onwards and in a three speaker system from 5 KHz onwards. • Also,
there is a super tweeter, which covers the range from 8 KHz onwards. • A
tweeter may be a small cone permanent magnet speaker or an
electrostatic type.
. * What is Crossover network • Audio crossovers are a class of electronic
filter used in audio applications. • Most individual loudspeaker drivers are
incapable of covering the entire audio spectrum from low frequencies to
high frequencies with acceptable relative volume and lack of distortion so
most hi‐fi speaker systems use a combination of multiple loudspeakers
drivers, each catering to a different frequency band.
. What is Crossover network • Crossovers split the audio signal into
separate frequency bands that can be separately routed to loudspeakers
optimized for those bands.
. The specific purpose of crossover network is: • To extend the frequency
range by the use of two or more speakers of different size. • To avoid inter
modulation distortion which may occur in a single unit. • To limit the input
to the most useful frequency range in a given speaker. • To protect a
delicate HF unit from LF input. • To facilitate suitable placing of bass and
treble speakers for natural results.
. 5) Horn Loud Speaker • A horn loudspeaker is a loudspeaker or
loudspeaker element which uses a horn to increase the overall efficiency
of the driving element, typically a diaphragm driven by an electromagnet. •
The horn itself is a passive component and does not amplify the sound
from the driving element as such, but rather improves the coupling
efficiency between the speaker driver and the air.
. Horn Speakers
. Horn Loud Speaker • The horn can be thought of as an "acoustic
transformer" that provides impedance matching between the relatively
dense diaphragm material and the air of low density. • The result is
greater acoustic output from a given driver. • Horns have been used to
extend the low frequency limit of a speaker driver.
. Horn Loud Speaker • When mated to a horn, a speaker driver is able to
reproduce lower tones more strongly. • The flare rate and the mouth size
determine the low frequency limit. • The throat size is more of a design
choice. • Horns have been known to extend the frequency range of a driver
beyond five octaves.
. Horn speakers • A horn facilitates the transfer of electrical energy into
acoustical energy and, if properly designed will be so with a minimum of
distortion. • The design of loudspeaker horn is complex and requires
careful consideration to prevent reflection of the acoustical energy back
into the horn bell.
. Horn speakers • The area of the throat determines the loading on the
diaphragm. • If the area of the throat is small compared to the area of the
diaphragm, the efficiency is increased because of the heavier loading
effect. • However, small throats require a longer horn, which increases the
frictional losses.
. Horn speakers • The reflex loudspeaker or bullhorn, a type of folded horn
speaker used widely in public address systems. • To reduce the size of the
horn, the sound follows in zigzag path through exponentially‐expanding
concentric ducts in the central projection (b, c), emerging from the outer
horn (d).
. Speaker Impedance is the impedance offered by a loud speaker at 400
Hz. The impedance will be changed with the frequency.
. Conference System It consists of Chairman unit, delegate unit and
secretary unit