Sound Reinforcement System

SUBMITTED TO NOWROSJEE WADIA COLLEGE AFFILIATED

TO SAVITRIBAI PHULE PUNE UNIVERSITY FOR THE
FULLFILMENT OF DEGREE OF
B. Sc. PHYSICS

SUBMITTED BY

Mr. Mohit Singh Chahar

Under the Guidance of

Prof. S. R. Kate
NOWROSJEE WADIA COLLEGE, PUNE
 CERTIFICATE

NOWROSJEE WADIA COLLEGE PUNE 411001

DEPARTMENT OF PHYSICS.

This is to certify that Mr. Mohit Singh Chahar has completed the
project work entittled

SOUND REINFORCEMENT SYSTEM

Which is being submitted to the department of physics, nowrosjee
wadia college ,pune -1 was carried out by the candidate under my
guidance at the DEPARTMENT OF PHYSICS, NOWROSJEE
WADIA COLLEGE PUNE -1

Prof. S. R. Kate Dr. F. I. Surve

(Project Guide) (Head of department)
 Acknowledgement

Words are often less to express one‟s deep regards. With an understanding
that work like this can never be outcome of single person, I take my opportunity to
express my profound sense of gratitude and respect to all those who have helped
me through the duration of this work.

I am grateful to my guide Prof S.R. Kate for his encouragement invaluable

guidance and supervision which inspired me to submit this report in the present
form

I would like to give my sincere thanks to Head Of Department Dr. F. I. Surve

for his valuable guidance and kind suggestion. I am very thankful to him for his
untiring devotion and inevitable assistance for development of this project

I would also like to express my gratitude to my teachers and seniors for their
valuable guidance.
Index

CHAPTER 1
Introduction to Sound Reinforcement System.
Conceptual Model.

CHAPTER 2
Practical Model.
Feedback Control.
CHAPTER 3
Working, Frequency Response and Acoustical Force.
Effects.

CHAPTER 4
Microphone.
Loudspeakers.

CHAPTER 5
Applications Of Sound Reinforcement System.
CHAPTER 1

This Chapter gives Brief introduction of Sound Reinforcement

System.

INTRODUCTION

A sound reinforcement system is the combination of microphone , signals

processer, amplifier and loudspeaker in enclosure all controlled by a mixing
console that makes live or pre-recorded sound louder and that also distribute those
sound to a larger or more distant audience
Traditionally a sound system is comprised of 3 basic component source,
amplification and speakers. This can be broken into more elemental pieces but if
one can have some version of these three, one can make sound at home
A Sound reinforcement system designed and engineered to make capture and
amplify sound and deliver it to an audience. The clear role it has to distribute sound
evenly to the areas where the listeners are, while ensuring that sound is not directed
to non listener areas such as wallas and seilings
The live sound reinforcement is a process in which an operator uses audio
technology to enhance and redistribute selected sound

Basic functions of a sound reinforcement system is

1) To help people hear something better

2) To enable sound lauder for artistic reason
3) To enable people hear sound in remote locations
 Conceptual model of a sound reinforcement system :

Acoustic to Electrical Electrical to

electrical energy manipulation of acoustical energy
conversation the Audio signals conversation

Sound field Sound field

( input source) (output source)
The input transducer (ie mic or pickup) concert sound into fluctuating electrical
current or any voltage is a precise representation of sound, the fluctuating current
or voltage is referred as an audio signal

The signal processing altars one or more characteristics of the audio signal. In the
simplest case it increase the power of the signal (a signal processer that do this is
called as amplifier)
In practical sound reinforcement system this block of the diagram represents a
multiple of device, pre amplifier, mixers, effect unit, power amplifier and so on
The output transducer (i.e output speaker or headphones) covert amplified or
processed audio signal back to sound
CHAPTER 2

This Chapter gives Brief information of Practical model of sound

reinforcement And Feedback Control.

Practical model of sound reinforcement

A system is designed to amplify the voice of three panelist. The system can be
analyzed as having three section
They are as follows

1. The input transducer

2. Signal processing
3. Output transducer
The three microphones are connected to individual input on the mixing cconsole

1 Preamplification the console microphone input section amplifies the level of

Audio signal from each microphone bringing it upto line level

2 Equalization The console provide means to adjust the tonal balance of each
microphone individually

3 Mixing The console adds the equalized signal of microphone together to

produce a single line level output signal

The environment is an Integral part of the sound system and it‟s effects must be
considered when the system is installed
Diagram :

Fig. Simplified Sound Model.

Feedback Control :

With the system powered up, by advancing the gain of the amplifier we will reach
a point where the system starts to ring. This ringing is called feedback.
Some of the sound from the loudspeaker is picked up by the microphone and re-
injected into the system, forming a continuous loop. Feedback occurs when the
gain in yhe loop reaches unity (gain of 0 dB).The feedback locks onto a frequency
for Which the loop path is non-inverting (in Phase).
If we turn the gain down so that the feedback just stops (loop gain just under
unity), the system‟s frequency response will still be erratic. This is because when
the loop gain is near unity, the system still resonates at those frequencies for which
the loop path is in phase.

The best approach to feedback control :

The best approach to feedback control is

A) Use directional elements (properly placed and aimed)

B) Keep the loudspeaker as far away from the microphone as practical (which is
one reason why many loudspeaker systems are flown high above the stage)

C) Keep the microphone close to the source (which is why many vocal performers
seem to be swallowing their mics).
Note :
Using directional elements in a sound system can increase maximum available gain
before feedback.
In sound recording and reproduction, equalization is the process commonly used to
alter the frequency response of an audio system using linear filters. … Since
equalizers adjust the amplitude of audio signals at particular frequencies, they
are,in other words frequency-specific volume knobs.
 Fig. Equilizer

Fig. Equilizer

Explaination :
In sound recording and reproduction, equalization is the process commonly used to
alter the frequency response of an audio system using linear filters. Most hi-fi
equipment uses relatively simple filters to make bass and treble adjustments.
Graphic and parametric equalizers have much more flexibility in tailoring the
frequency content of an audio signal.
Equalizers are used in recording studios, radio studios and production control
rooms, and live sound reinforcement and in instrument amplifiers, such as guitar
amplifiers, to correct or adjust the response of microphones, instrument pick-ups,
loudspeakers, and hall acoustics. Equalization may also be used to eliminate or
reduce unwanted sounds (e.g., low hum coming from a guitar amplifier), make
certain instruments or voices more (or less) prominent, enhance particular aspects
of an instrument‟s tone, or combat feedback (howling) in a public address system.
Equalizers are also used in music production to adjust the timbre of individual
instruments and voices by adjusting their frequency content and to fit individual
instruments within the overall frequency spectrum of the mix.
Equalization is commonly used to increase the „depth‟ of a mix, creating the
impression that some sounds in a mono or stereo mix are farther or closer than
others, relatively.
Equalization is also commonly used to give tracks with similar frequency
components complementary spectral contours, known as mirrored equalization.
Select components of parts which would otherwise compete, such as bass guitar
and kick drum, are boosted in one part and cut in the other, and vice versa, so that
they both stand out.
CHAPTER 3

This Chapter gives Brief information of Working ,frequency

response and Acoustical force And Effects in Sound Reinforcement.

Working ,frequency response and Acoustical force :

Diagram :

Explaination :
A practical audio system is a network of many different types of devices
through which the signal must pass on its way to the listeners' ears. Each stage in
the system will have specific frequency response characteristics, and will therefore
modify the signal to some degree. The overall frequency response of the system is
a function of the combined responses of all the various elements in the chain.The
audio elements that exhibit the flattest frequency response characteristics are
electronic circuits and cables.
standard octave or l/a-octave spacing, it will coincide with the bands measured on
an ISO standard Real Time Analyzer. Instead of a sine wave, the excitation signal
used for measurements like this is a special signal called pink noise. Pink noise is a
randomly generated signal that excites all the audio frequencies with equal energy
per octave. It sounds very much like a rushing waterfall.

It is important to note that measurements like this can conceal as much as they
reveal. While they provide a good picture ofthe general characteristic of a device's
frequency response, very narrow peaks or dips may be completely missed by the
technique. In a reverberant space, the measurement may say more about the room
than about the loudspeaker under test.

Effects :
The frequency response of any sound system is affected by the environment
in which it works. Outdoors, the main factors affecting the system's response are
wind, temperature, and air absorption. Wind tends to deflect sound very slightly as
it travels, and gusty wind can appear to modulate the sound. Temperature gradients
can also deflect sound, and to an even greater degree than wind gradients. Air
absorption affects mainly high frequencies. As sound travels through air, the
energy of the highs is lost more quickly than that of the lows. That is why sounds
heard at a distance appear muffled. The degree to which the air absorbs high
frequency energy is affected by the relative humidity. Indoors, the predominant
factors affecting a system's response are reflections from wall, ceiling and floor
surfaces, and room resonances. Reflections not only cause reverberation effects,
but also can cause cancellations at specific frequencies which show up as dips in
the system response. Room resonances can cause dips or, more commonly, peaks
in the response. All these effects will color the sound of the system.
The fundamental there are
Additional sine wave components
Whose frequencies are multiples of the Fundamental frequency.
Mental is at 500 Hz, for example, the

Harmonics will occur at 1 kHz, 1.5 kHz, 2 kHz, 2.5 kHz, and so on.

• Wind effect
Wind effects are divided into two classes - velocity effects and gradient effects. A
crosswind will add a velocity vector to a propagating sound wave and can shift the
direction of propagation of the sound, making it appear to come from a different
location.Wind velocity gradient effects occur when one air layer is moving at a
different speed than an adjacent layer usually one layer is above the other.Such a
gradient might be encountered when the audience area is shielded from the wind by
a barrier, such as a stand of trees or a wall.
Sound propagated outdoors is subject to the influence of environmental factors that
are not significantly present indoors. Such influences can cause the behavior of
sound systems to deviate from that predicted by inverse square calculations.The
principal factors affecting sound outdoors are wind, temperature gradients, and
humidity. The effects of these factors are most noticeable in large-scale outdoor
events, such as sports events or rock concerts.

Fig. Explaining Wind Effect

CHAPTER 4

This Chapter gives Brief information of Types of Microphones And

Loudspeakers.

Microphone :
Introduction. In the preceding chapter we have considered two important types of
electroacoustic transducers which are capable of converting electrical energy into
acoustic energy. Of equal importance are those electroacoustic transducers used to
convert acoustic energy into electrical energy. When operating in air such
transducers are known as microphones, and when operating in water as
hydrophones. Microphones. serve two principal purposes. First, they are used for
converting music or speech into electrical signals which are transmitted or
processed in some manner and then reproduced. Second, they serve as measuring
instru ments, converting acoustic signals into electrical currents which actuate
indicating meters.

Numerous physical phenomena have been used to convert acoustic energy into
electrical energy. These include electromagnetic induction, the piezoelectric effect,
magnetostriction, variations in the capacitance of a capacitor, and variations in the
resistance of packed carbon granules. Before the development of the vacuum-tube
amplifier, the inherent insensitiveness of all but the last of these methods precluded
their practical application to sound systems, and all such systems employed carbon
microphones. At present, however, the voltage and power gains that can be
obtained from vacuum-tube amplifiers make it possible to employ microphones of
much lower sensitivity, such as electrodynamic micro phones, crystal microphones,
and condenser microphones, and thus to take advantage of the greater uniformity of
response and the absence of internal noise that is characteristic of these types.

All microphones are used to convert the periodic variations of acoustic pressure in
the medium into similar variations in voltage or current in an associated electrical
circuit If this electrical response corresponds to the variations in acoustic pressure,
the microphone is classified as a pressure microphone; if the response corresponds
to variations in the pressure gradient, it is known as a pressure-gradient
microphone Microphones may also be classified as either sound-powered or sound-
controlled In sound-powered microphones the acoustic energy of the incident wave
supplies the electrical energy generated in the microphone; in sound controlled
microphones the acoustic waves merely control the flow of electrical energy from a
battery or other source of electrical power.

11.2 Carbon Microphone.

The carbon microphone is widely used for telephone and radio communication
purposes where its high electrical output, low cost, and durability are of greater
significance than fidelity of response. It depends for its operation on the variation
in resistance of a small enclosure filled with carbon granules, which is known as
the carbon button. Figure 11.la gives a schematic diagram of this type of
microphone. As the diaphragm is displaced, the plunger attached to it varies the
pressure applied to the carbon granules, and hence the resistance from granule to
granule, so that the total resistance across the carbon button, which is ordinarily
about 100 ohms, varies in an approximately linear manner with the acoustic
pressure applied to the diaphragm.

In the simple equivalent circuit shown in Fig. 11.1b, let us assume that for small
displacements of the diaphragm the resistance R, of the carbon button varies
linearly with the displacement y of the center of the dia phragm, i.e.,
Rc = Rc + hy
where Ro is the zero-displacement resistance of the button and his its resistance
constant in ohms per meter of displacement of the plunger. At frequencies well
below the fundamental resonant frequency of the dia phragm, its motion is
stiffness-controlled, so that it may be treated as a simple oscillator of stiffness r and
negligible mass. If an impinging sound wave of frequency to produces a pressure
amplitude P on the face of the diaphragm, then
where S is the effective area of the diaphragm and y, is the displacement amplitude
at its center. Therefore
Rc = R0 + hy0

Fig. (a) Simple Carbon Microphone. (b) Equivalent Electrical Circuit

This variation in resistance causes the current in the circuit to vary as

(11.4)

where E, is the voltage of the battery. If hy0 < R0 then equation 11.4 may be
expanded and simplified to
Equation 11.5 indicates the presence of a steady direct current E0/R0, an alternating
current ic = -(E0hy0 / R02) cos wt, and higher harmonics of this current. The
alternating current component i, of frequency oo may be thought of as arising from
an alternating emf

generated internally in the microphone. This treatment is somewhat analogous to

that in which the effect of a grid voltage e, applied to a vacuum tube is replaced by
an equivalent voltage -ue, generated in the plate circuit. The amplitude of the
equivalent voltage of the microphone is therefore

The ratio Ec/P is a measure of the sensitivity of the microphone and is known as the
open-circuit voltage response. It may be expressed either in volts per newton/m,
i.e.

or upon multiplying the right-hand side of equation 11.8 by 0.1, it may be

expressed in volts per microbar. The response may also be expressed as a decibel
level relative to some arbitrary reference level. The commonly used reference level
is one volt per microbar (one volt per 0.1 newton/m³). Expressed in terms of this
reference level, the db response of the above carbon microphone is

The response evidently increases as the battery voltage E0, is increased or as the
total resistance R0, of the circuit is decreased. It also increases directly with the
area S and inversely with the stiffness s of the diaphragm.
 Typical constant-pressure frequency-response characteristic of a carbon
microphone in decibels relative to 1 voit per microbar.

However, there are practical limitations to such methods of increasing the

sensitivity, since high values of E, cause excessive heating and internal noise in the
carbon button, and decreasing s tends to lower the funda mental frequency of the
diaphragm, thus reducing the useful frequency range of the microphone.

The ratio of second-harmonic voltage to fundamental voltage, as obtained from

equation 11.5, is hy/2Ro Since increasing the ratio h/R, in an attempt to increase
the sensitivity also increases the relative amplitude of the second harmonic, it is
apparent that some sensitivity must be sacrificed if distortion is to be reduced, and
vice versa. It is also apparent that very intense sound waves, for which the
amplitude y, is large, will introduce considerable harmonic distortion into the
output of a carbon microphone. Finally, the assumption that h is constant, i.e., that
the resistance varies uniformly with the displacement, is undoubtedly in error for
large amplitudes and will act as a further source of distortion. The distortion arising
from even-harmonic terms in equation 11.5 can be balanced out by using a double-
button carbon microphone connected in a push-pull type of circuit, but this type of
microphone will still have odd-harmonic distortion when the amplitude of the
impressed sound wave is too great.

A measured frequency-response characteristic for a single-button carbon

microphone is shown in Fig. 11.2, which gives the open-circuit response n in
decibels as a function of frequency. In this particular response curve, which is
known as a constant-pressure curve, the pressure actuating the diaphragm is
uniformly distributed over its surface and is measured directly. When a
microphone is inserted into a sound field the sound pressure acting on the
microphone is not necessarily equal to that of the undisturbed sound wave, for the
presence of the microphone will produce diffraction effects which may either
increase or decrease the pressure as compared with that in the free wave. Response
curves showing the ratio
of the open-circuit voltage to the pressure in the undisturbed sound wave are
known as free-field curves; they will be considered further in Sect. 11.7.

The peak response in Fig. 11.2, which occurs near 2000 cycles/sec, is associated
with the fundamental resonant frequency of the diaphragm, and the uneven
response above this frequency is due to the breaking up of the diaphragm into
various overtone modes of vibration. If the diaphragm is tightly stretched, instead
of being merely clamped at its rim, its effective stiffness can be increased, with a
corresponding increase in the funda mental frequency. The use of a stretched
diaphragm makes it possible to extend the region of relatively uniform response to
about 8000 cycles/sec, but only at the expense of a decreased sensitivity.

In using a carbon microphone it is necessary to insert a load Z, into the simple

circuit of Fig. 11.1b. An efficient method of obtaining maximum output is to
couple the micro phone through a transformer, as shown in Fig. 11.3. Here the
signal voltage E, developed in the primary of the transformer is
As is to be expected, the maximum power is developed in this load when the
impedance Z, ia pure resistance of magnitude R0

Condenser Microphone
A condenser microphone is one that depends for its operation on the variation in
capacitance between a fixed plate and a tightly stretched metal diaphragm. Its
development by Wente in 1917 represents an important milestone in the history of
modern electroacoustics, and for a number of years this type of microphone was
the accepted standard for high-quality sound systems. However, the condenser
microphone has a number of disadvantages, such as a very high internal
impedance, which necessitates locating a preamplifier in the immediate vicinity of
the microphone and leads to the generation of noise in the high-impedance circuit
required to couple it to the grid of the preamplifier..tube. The microphone also
requires a polarizing voltage ranging from about 200 to 400 volts, which must be
supplied by batteries or by a rectifier having an exceptionally well-filtered output.
As a conse quence of these disadvantages, condenser microphones have been sup
planted by either crystal or electrodynamic microphones in many sound systems
but their extensive use as primary standards for calibration
purposes, in acoustic research, and for high-fidelity recording of sound, still
justifies a somewhat detailed analysis. The simple condenser microphone shown in
Fig. 11.4a consists of a

thin stretched metal diaphragm, usually of steel or aluminum, having a radius a,

and separated by a small distanced from a parallel rigid plate. The rigid plate is
insulated from the remainder of the microphone, and a polarizing voltage E0, is
applied between it and the diaphragm, has indicated in the accompanying circuit
diagram. When a sound wave impinges upon the diaphragm, the resulting
displacement of the latter alters the electrical capacitance C of the microphone,
causing a signal voltage eL to appear across the load resistor R Let us assume that
the capacitance at any instant is given by

where C0, is the capacitance in the absence of any applied pressure, and C1, is the
amplitude of the change in capacitance resulting from the appli cation of a
sinusoidal pressure variation. In the circuit of Fig. 11.4b

Substitution of equation 11.10 into equation 11.11 and differentiation with respect
to time t gives

A series solution of this equation is most readily obtained by assuming

In all practical condenser microphones C₁ < Co, even for very intense sounds, and
consequently the amplitudes A2 A3. A4 …. of the higher harmonic overtones are
negligible compared to A1 then

The voltage drop across the resistor Ry, resulting from this current is

Equation 11.15 indicates that a condenser microphone may be considered as

equivalent to a generator having an open-circuit voltage amplitude

and an internal capacitive impedance 1/jwC0 When the radius a and distance d are
expressed in meters, the capacitance C0, of the microphone in farads is given by

where €0 = 8.85 x 1012 farad/meter is the permittivity of free space. Upon

multiplying by 10 this equation becomes

which gives the capacitance in µµf. Let us now assume that the pressures in the
sound wave displace the stretched diaphragm in a manner analogous to that of the
forced vibrations of a membrane, as discussed in Sect. 4.8. Then for such low
driving frequencies that ka < 1, or, as given by equation

equation 4.34 shows that the average displacement y of the surface of the
diaphragm is

In this equation P represents the pressure amplitude in newtons per square meter,
and 7 the tension in the diaphragm in newtons per meter. Dis placing the
diaphragm of the microphone an average distance Ӯ from its
normal position towards the fixed plate will change its capacitance from C0 to C,
where

Since the amplitude of Ӯ is always small compared to d, equation 11.19 may be

expanded as a series

Neglecting (Ӯ /d)2 and higher-order terms, this becomes

A comparison of this equation with the previously assumed equation

so that the theoretical amplitude of the open-circuit voltage of this idealized simple
condenser microphone, as defined by equation 11.16, is

3.] Moving coil Electrodynamic microphone :

The simple moving-coil or "dynamic" microphone consists of a light diaphragm, to
which at small coil is rigidly attached. The action of sound waves on the
diaphragm. causes the coil to move in the radial field of a permanent magnet, thus
generating an emf expressed in volts of
e = Blv
where B is the flux density of the magnetic field in webers/m2, / is the length of
wire in the coil expressed in meters, and is its velocity in meters per second.
Basically, a moving-coil microphone is similar to a direct radiator loudspeaker,
except that it converts acoustic energy into electrical energy, instead of the reverse.
As a matter of fact, the small direct radiator loudspeakers of an interoffice
communications system usually serve as the microphones of the system
As an introduction to the problem of designing a satisfactory moving coil
microphone, let us consider its mechanical system as equivalent to a simple
oscillator of mass mo, stiffness So, and resistance Pv The velocity of the coil is
then

where S is the area of the diaphragm and P is the pressure amplitude acting on the
diaphragm. Consequently, if the voltage output for constant sound pressure on the
diaphragm is to be independent of frequency, the mechanical impedance Z must
also be independent of frequency. This requires that R, be large, i.e., that the
system be resistance-controlled. On the other hand, a high voltage sensitivity can
be obtained only by making the velocity amplitude as large as possible, which
requires a small mechanical impedance. Therefore, it is impossible to satisfy both
the requirement of high sensitivity and uniformity of response by employing a
mechanical system equivalent to a simple oscillator. Instead, additional mechanical
elements are utilized to compensate for the increase in mass reactance com, at high
frequencies and in stiffness reactance s/w at low frequencies, and thus to provide a
fairly uniform response, and yet a low mechanical impedance.

It is possible to design electrical networks capable of transmitting wide bands of

frequencies by combining a number of resonant circuits. In 1931 Wente and
Thuras¹ published a paper showing how the same expedient could be resorted to in
the design of a satisfactory mechanical system for the moving-coil microphone.
The success of their efforts is but one of many examples illustrating the fruitfulness
of the application of electrical network techniques to acoustic problems.
Let us consider the mechanical circuit of Fig. 11.9. It can be shown that
the input impedance is..

The mechanical constants Ro, ma, F. R₁, m, and s, can be chosen so that the
absolute value of the mechanical impedance Z, is fairly uniform over a rather wide
range of frequencies Curve 4 of Fig. 11.10 shows its value in the range of
frequencies from 40 to 10,000 cycles/sec for the following Values of the
mechanical constants : R₁=1, R₁ = 24, 5, 10,000, 5= 1.000,000, m, 0.0006 kg, and
m of a 0.0003 kg. For purposes of comparison, the mechanical impedance of the
simple oscillator RmmS alone has been plotted in curve B, and that of a similar
oscillator for which R, is 25 in curve C. It is evident that curve A is flatter than
either curve B or curve
C, and consequently the corresponding mechanical system will lead to a more
uniform response of the microphone. Upon combining equation 11.23 with
equation 11.24 the expression for the open-circuit response M of a moving-coil
microphone expressed in volts per newton/m² becomes
A cross-sectional view of a moving-coil microphone having the mechanical char
acteristics of the circuit of Fig. 11.9 is shown in Fig. 11.12. The diaphragm is
composed of a spherical shell which vibrates as a rigid piston in the useful range of
frequencies. The stiffness s, and the resistance R, are contributed by the corrugated
annulus supporting the dia phragm. The stiffness s, is due primarily to compression
of the air trapped in the Figs. . The of a neoving-coil microphone. mass m, and the
resistance R, arise from viscous forces opposing the flow of air through the
capillaries of the si cloth. The stiffness due to the air chamber below the silk cloth
is relative small and may be neglected.

Loudspeakers

Introduction.
In, practically all, modern acoustical work, the oscillations in sound pressure are
picked up by, some form of receiving electroacoustic transducer, e.g., a
microphone, which converts them into similar electrical current or voltage
oscillations, The latter either may be electrically amplified for immediate
reconversion into sound energy or may be stored in some form, such as on
magnetic tape, for later analysis or playback. After amplification, these
oscillations, may be converted, back into sound vibrations by some form of
transmitting electroacoustic
transducer, such as the so-called loudspeaker (speaker)./ There are a number of
interrelated factors that must be considered in the design, of an effective transducer
for converting electrical energy into airborne acoustic energy, These include
electroacoustic efficiency, uni formity of frequency response, linearity of
amplitude response, transient response, power handling capacity, size, durability,
and cost. An ideal loudspeaker:

(1) Would have an electroacoustic efficiency appro..ching 100

per cent. (2) Would have an acoustic output response that is independent
of frequency over the entire audible range.
(3) Would introduce neither harmonic nor intermodulation: distortion into its
output.
(4) Would faithfully reproduce transients, as well as steady input signals.
(5) Would be capable of producing a nondirectional radiation
pattern (6) Would be of as small a size as is possible considering the required
acoustic output.
No single transducer, has been designed that is capable of satisfying all the above
requirenients of the many devices developed for the radiation of acoustic energy
into air, the two most widely used are the direct-radiator
or dynamic loudspeaker and the horn loudspeaker. Both of these loud speakerr
utilize the electrodynamic coupling that exists between the motion of a vibrating
surface, called the speaker cone or diaphragm, and the current in a so-called coice-
coil, Additional types of electromechanical coupling that are used for this purpose
include electrostatic coupling in electrostatic loudspeakers and electromagnetic
coupling in telephone. receivers. In this chapter, we will concern our selves
primarily with the characteristics of direct radiator loudspeakers and to a lesser
extent with those of horn loudspeakers.

Idealized Direct Radiator Speaker.

A loudspeaker that doesn‟t have a horn between the moving element and the air.
Often direct radiator-type speakers are designed for home use, while horn-type
speakers are preferred for sound reinforcement applications. Direct radiators
generally provide smoother, more uniform response, while horns are much more
efficient, providing a greater output level for a given power input. Also, horns have
greater directivity, which is usually desirable in sound reinforcement systems.

Efficiency Equation :-

Where = Bl is transformation factor.

Rm = mechanical resistance
Rr = radiation resistance
RE= electrical resistance
Zm= mechanical impedance.

2.) Equivalent Circuit :

Acoustic Power output in watts

Mechanical Resonance = Xr + W0m – s/W0 = 0
W0 = 2 f0

Horn Loudspeaker
A horn loudspeaker is a loudspeaker or loudspeaker element which uses
an acoustic horn to increase the overall efficiency of the driving element(s). A
common form (right) consists of a compression driver which produces sound
waves with a small metal diaphragm vibrated by an electromagnet, attached to a
horn, a flaring duct to conduct the sound waves to the open air. Another type is
a woofer driver mounted in a loudspeaker enclosure which is divided by internal
partitions to form a zigzag flaring duct which functions as a horn; this type is
called a folded horn speaker. The horn serves to improve the coupling efficiency
between the speaker driver and the air. The horn can be thought of as an
"acoustic transformer" that provides impedance matching between the
relatively dense diaphragm material and the less-dense air.

There are 3 types of Horn Loudspeakers

1. Conical
2. Exponential

3. Hyperbolic
For exponential horns let So is the cross sectional area of the throat then :-
Sx = Soemx
m is called as flare
Sx is the cross sectional area at the distance x
Aerodynamic Resistance is high near throat and low near mouth
Horn will have :-
Profile
Sx = Soemx
Its behaviour is govern by the wave equation ;-

Cutoff frequency ;- this frequency is defined as the one below which the horn
loudspeaker doesnot respond.
Fc=mct/4ԥ
m – is the flare of the horn
Ct – is the velocity of sound in air at temperature t .
Efficiency of horn loudspeaker is given by the equation given below :-
CHAPTER 5

This Chapter gives The Practical Application of Sound

Reinforcement in Everyday Life.

Applications :
Signal sources are an important component in many test systems and are designed
to provide a source of “clean” or near ideal signals so that distortions do not
influence the test results. Signal sources are also used to simulate real-world
stimuli to test a device‟s performance in a realistic environment

Tape recorder :
An audio tape recorder, also known as a tape deck, tape player or tape machine or
simply a tape recorder, is a sound recording and reproduction device that records
and plays back sounds usually using magnetic tape for storage.
In its present-day form, it records a fluctuating signal by moving the tape across a
tape head that polarizes the magnetic domains in the tape in proportion to the audio
signal. Tape-
recording devices include the reel-to-reel tape deck and the cassette deck, which
uses a cassette for storage.

Phonograph records :

A phonograph disc record (also known as a gramophone disc record, especially in

British English), or simply a phonograph record, gramophone record, disc record
or record, is an analog sound storage medium in the form of a flat disc with an
inscribed, modulated spiral groove. The groove usually starts near the periphery
and ends near the center of the disc.

The phonograph recorded and stored sound mechanically by etching sound waves
(or more accurately, the electrical signal of the sound waves) with a needle, onto
tinfoil cylinder. The cylinder was rotated by a hand crank and the needle moved to
cut a groove into the tinfoil, recording the sound wave signal.
Broadcasting signals :
The broadcast of a single signal, such as a monophonic audio signal, can be done
by straightforward amplitude modulation or frequency modulation.... In television
transmission, three signals must be sent on the carrier: the audio, picture intensity,
and picture chrominance.
Compact disc :
To listen to music using a CD player with a headphone output jack, the user plugs
headphones or earphones into the headphone jack.
Compact disc audio (CDA) is the standard format for audio compact discs (CD).
The CD format is the same that is used in regular CD players
 Bibliography and References:

1.Lawrence E. Kinsler, Austin R. Frey, Alan B. James V. SandersFundamental of

Acoustics , 4th Edition

2.Dr.Farhat Surve –Notes :

1.) microphones
2.)loudspeakers
3.)sound reinforcement systems.

3.Sites.google.com/site/worldacoustics