MODULE 5

PROPOGATION OF SOUND
Sound is produced by a vibrating body. A material medium is necessary for
the propagation of sound waves. In wave motion momentum and energy are
transferred. Characteristics of wave motion.
1. Wave motion is a disturbance produced in the medium by the repeated
periodic motion of the particles of the medium. .
2. Only the wave travels forward whereas the particles of the medium
vibrate about their mean positions.
3. There is a regular phase change between the various particles of the
medium. The particle ahead starts vibrating a little later than the
particles preceding it.
4. The velocity of the wave is different from the velocity with which the
particles of the medium are vibrating about their mean positions. The
wave travels with a uniform velocity whereas the velocity of the
particles is different at different positions. It is maximum at the mean
position and zero at the extreme positions of the particles. There are
two types of wave motions.
a. Transverse wave.
b. Longitudinal wave
Sound wave are longitudinal waves and light waves are transverse waves.
Figure 1 shows the formation and propagation of transverse and longitudinal wave.
A sound wave is propagated and conveyed to the ear by means of the
intervening layers of air. Consider a vibrating tuning fork. Let us confine our
attention to the right hand prong only. When it moves towards the right, it
compresses the layer of air in front of it and as a consequence the pressure of this
layer will be greater than the adjacent layers. It tends to relieve the strain thus
created, by compressing them. These in turn hand on the compression. Thus a pulse
of compression will travel onwards to the right. Again, when the prong moves
towards the left rarefaction is started. These follow one another and as the fork
vibrates, compressions and rarefaction are sent out in regular succession. These
waves at last reach the car and set the tympanic membrane, which is ultimately
transmitted via a system of bones and cords to the brain and causes the mental
sensation called sound.
MEASUREMENT OF INTENSITY OF SOUND
The intensity of sound is defined as the quantity or energy propagating
through a unit area per unit time, the direction of propagation being perpendicular to
the area. The amount of power transmitted is measured in Watts/m
2
. A convenient
unit is microwatts /m
2
.
According to Weber Fechner law in psychology, the loudness of a sound as
judged by the ear is proportional to the logarithm of the intensity. If l
1
and l
0

represents the intensities of two sounds of a particular frequency and L
1
and L
0
the
corresponding measure of loudness then L
1
= K log l
1
and L
0
= K log 1
0
. The
difference in loudness technically known as Intensity Level, L between them is given
by
L = L
1
- L
2
= K[log
1
- log 1
0
]
1
2
l
L Klog( )
l
=

Where K is a constant that depends on the units and I
0
is some standard
reference intensity arbitrarily taken as 10
-12
watt/m
2
, which corresponds to the
intensity of the sound which can be just heard at a frequency of about 1000 cycles
per second. This is threshold of audibility of a normal ear.
Where K in the equation above is taken as I the difference in loudness is
expressed in bels, a unit named in honour of Alexander Graham Bell, the inventor of
telephone. This unit of loudness is rather too large, one tenth of it, the decibel (db)
has become the standard. So in order to express the difference in loudness of a sound
of intensity I in decibel, the above relation should be written as
1
2
l
L 10log( )
l
=

To build a scale of loudness, we have to fix its zero. The loudness
corresponding to the threshold of hearing is the zero of this scale. This occurs when
the intensity of sound wave equals 1
0
or 10
-12
watt/m
2
. The maximum intensity which
the ear can tolerate without sensation of pain is about 10
-2
watt/m
2
and it corresponds
to the intensity level
2
12
10
L 10log( ) 120db
10

= =
The following table gives the approximate value of some sound measured in
decibels.
Source Intensity level in Decibel
Threshold of hearing 0
Rustle of leaves 10
Whisper 15-20
Ordinary conversation 60-65
Motor trucks and heavy street traffic 70-80
Roaring lion at 20 ft. 90
Thunder 100-110
Painful sounds 130 or above

In the above table we have expressed the loudness in decibels on the
assumption that the threshold of audibility is the same for all frequencies of the ear
and the limits of audibility vary over wide ranges of intensity and frequency, hence
the sound of same intensity but different frequencies seem to differ in loudness.
Therefore a different unit for measuring loudness is used. It is called the phon. The
measure of loudness in phons of any sound is equal to the loudness in decibels of an
equally loud pure tone of frequency 1000 cycles/second.
Acceptable noise levels:
Type of residential Area Acceptable noise level in dB
Rural 25-35
Suburban 30-40
Residential Urban 35-45
City 45-55
Industrial area 50-65
Outdoor noise levels in residential areas
Type of place/building Acceptable noise level in dB
Radio and TV studios 25-30
Music room 30-35
Hospital, classroom, Auditorium 35-40
Apartments, hotel, home 35-40
Conference Room, small office, concert
room
40-45
Private offices 40-45
Libraries, Large public office, banks,
stores
45-50
Restaurants 50-55
Indoor noise levels in public/private places
The above table is as per IS standards.
AIR COLUMNS
Stationary longitudinal waves can be produced in a column of air by any
periodical movement whose frequency coincides with one of the natural frequencies
of the column. All wind instruments are provided with a column of air called a
resonator, which may be in the form of a rectangular air chamber. The periodic
movement is caused by an important part of the musical instrument called the mouth-
piece, which is different in construction in different instruments: It is the mouth -
piece that acts as a generator and supplies the energy necessary to maintain the
vibrations in the column of air. In the theoretical treatment the following assumptions
are made:
1. The diameter of the pipe is small compared with the length of the pipe
and with the wave length of sound.
2. The diameter is sufficiently great so that the viscosity effects can be
neglected.
3. The walls of the pipe are rigid.
The organ pipes are classified into two groups: Flute or Flue pipes and Reed
pipes.


DOPPLER EFFECT
It is commonly observed that the pitch of a note apparently changes when
either the source or the observer are in motion relative to each other. When the
source approaches the observer or when the observer approaches the source or when
both approach each other the apparent pitch is higher than the actual pitch of the
sound produced by the source. Similarly when the source moves away from the
observer or when the observer moves away from the source or when both move away
from each other, the apparent pitch is lower than the actual pitch of the sound
produced by the source.
This apparent change in pitch due to relative motion between the source and
the observer is called Doppler effect.
Doppler effect in sound is asymmetric, when the source move towards the
observer with a certain velocity, the apparent pitch is different to the case when the
observer is moving towards the source with the same velocity. But it is not so in the
case of light. Doppler effect in light is symmetric.
The apparent pitch in different cases is calculated below
Let n - pith of sound
- wavelength
v - velocity of sound
n' - apparent pitch
Case 1: When the source moves towards the stationary observer with a
velocity a

v
n n
v a
| |
' =
|

\ .

Case 2: When the source moves away from the stationary observer with a
velocity a

v
n n
v a
| |
' =
|
+
\ .

Case 3: When the observer moves towards a stationary source with a
velocity b

v b
n n
v
+ | |
' =
|
\ .

Case 4: When the observer moves away from a stationary source.

v b
n n
v
| |
' =
|
\ .

Case 5: When the source moves towards the observer and the observer
moves away from the source.

v b
n n
v a
| |
' =
|

\ .

Case 6: When the source and the observer move towards each other

v b
n n
v a
+ | |
' =
|

\ .

Case 7: When the source and the observer move away from each other.

v b
n n
v a
| |
' =
|
+
\ .

Case 8: Source moving away from the observer and the observer moving
towards the source.

v b
n n
v a
| |
' =
|

\ .

MUSICAL SCALES:
CHORD: When two notes of different frequencies are sounded
simultaneously, their combination is called a chord. In the case of a concord or
consonance the combination produce a pleasant or agreeable effect. While other
combinations produce a disagreeable or unpleasant effect and it is called dissonance.
HARMONY: When the two notes sounded together produce concord their
combination is called Harmony.
MELODY: If the two notes are sounded one after the other, combination
is called melody.
DIATONIC MUSICAL SCALE: A series of notes separated by definite
and simple intervals so as to produce a musical effect when played in succession is
said to constitute a musical scale. The most common musical scale is the Gamut or
the Diatonic scale. It consists of a series of eight notes, the interval between last and
the first note being 2/1. It is therefore called an octave. The series of note is denoted
as:
C D E F G A B C
sa re ga ma pa dha ni (sa)'
All these notes are arranged in increasing order of frequencies so that they
present a regular graduation in pitch and their vibration frequencies are represented
by
l
9
8

5
4

4
3

3
2

5
3

15
8
2
l 1.125 1.25 1.333 1.500 1.667 1.875 2
i.e., if the frequencies of first note C called the tonic or the key note be taken
as 24, the relative frequencies (ratio of successive frequencies) of the various notes
of the diatonic scale will be
(24) (27) (30) (32) (36) (40) (45) (48)
9
8

10
9

16
15

9
8

10
9

9
8

16
15

(Major (Minor (Semi (Major (Minor (Major (Semi
tone) tone) tone) tone) tone) tone) tone)
If the frequencies of the note C be taken as 256 and 264 respectively, the
various notes of the scale will be denoted by
256 288 320 341.3 384 426.7 480 512
264 297 330 352 396 440 495 528
The above scale consists of three main intervals Major tone, Minor tone and
semi major tone respectively, so that the sequence of interval in Diatonic scale is
major tone, minor tone, semi-tone, or neglecting the difference between major and
minor tones, tone, tone, semi-tone, tone, tone, semi-tone. Since the major tones occur
more frequently, this scale is called Major Diatonic scale. It could be extended both
above C and below C level. Intervals within an octave with small intervals viz.,
4
3
,
3
2
, 2 are most consonant and are named as fourth, fifth and octave their names being
derived from the positions of these notes on the scale.
MICROPHONE
The microphone is essentially an arrangement for the conversion of sound
vibrations into vibrations of electrical current. In the telephone communication
system it is very successfully used as a transmitter and the vibrations of electrical
current thus produced are converted into sound by the receiver at the distant end. It is
also the first element of a loud speaking equipment or a broadcasting arrangement in
which electrical oscillations after proper amplification are reconverted into sound by
loud speaker. The general principle of modern carbon microphones is shown in fig.
It is based on the variation of the resistance of fine carbon granules when subjected
to pressure changes. Carbon granules are enclosed in between two plates one of
which is fixed and the other serves as a diaphragm which responds to rapid change in
pressure. The plates are placed in series with a key, a battery, and the primary of a
transformer, the secondary of the transformer is connected to the telephone receiver.
When the key is closed, a steady small current flows through the circuit. If R be
resistance of the circuit when there is no displacement of the diaphragm and Ka
sinet the varying resistance due to the displacement a sinet at any instant. K being
constant of the microphone then the total resistance of the circuit at any instant is
(R+Kasin et). Let V be the direct emf of the circuit, then the current at any instant is
given by
1
V V Ka
i l sin t
R Kasin t R R

(
= = + e
(
+ e


2 2
2
2
V Ka K a
l sin t sin t .......
R R R
(
e + e
(


The first term indicates a steady current when the diaphragm is at rest, the
second an alternating current of the same frequency as is impressed on the diaphragm
and the rest of the terms denote its other harmonics. The current is thus modulated.
This modulated electrical current passes through P, the primary of a transformer and
produces by induction a corresponding varying current in S, its secondary. This
amplified current passes through the telephone T and excites its diaphragm. The
movements of the latter set the air in corresponding vibrations reproducing the
original sound.Other commonly used types of microphones are
The Electrodynamic Microphone: This microphone is base don the
principle of electromagnetic induction consisting of a small coil of wire attached to
the back of a freely moving light plate. The coil is situated in the magnetic field
between the central pole piece and the peripheral pole piece of a permanent "pot"
magnet. The sound wave cause the plate end of the coil to vibrate and varying
current thus induced in the coil are amplified and conveyed to the distant end.


Ribbon or Velocity Microphone: It is also based on the principle of
electromagnetic induction.
The Condenser Microphone: It is based on the principal that a charged
condenser connected to an electrical circuit be subjected to sound vibrations, they
will produce variations in the distance between the plates thus changing its capacity,
then an alternating current will be set up in the circuit.
The Crystal Microphone: This microphone is based on the principle of
piezoelectric effect, according to which certain crystals like quartz, Rochelle Salt
produce minute differences of potential between its opposite faces when subjected to
pressure.
The Hot Wire Microphone: It is based on the principle that resistance of a
metallic wire changes with change in temperature.
THE LOUDSPEAKER:
It is a device for converting electrical energy into sound energy and
therefore essentially a microphone worked backwards. It is provided with a horn or a
circular board called 'baffle' for effectively transferring the vibrations of the moving
part to the external air. The commonest and the most efficient type of speakers
nowadays is the moving coil type. It is based on the principle that when a variable
current is passed through a conductor in a magnetic field, the conductor is acted on
by a variable force in accordance with Fleming's Left Hand Rule and is f the current
is oscillatory the conductor is set in vibration.
The moving part of the apparatus consists of a small coil called the 'voice
coil' wound on a cylindrical strip to which the variable current output of the
microphone is fed. The voice coil is free to move in the annular gap between the
central S and the peripheral pole piece N, of a 'pot' magnet designed to produce a
strong radial magnetic field in it. It is usually magnetized by a steady (DC) current
flowing in the coil wound round it. The coil is attached to a conical diaphragm made
of parchment with circular corrugation and supported round the periphery by a
flexible annular strip of leather or rubber. When the variable current passes through
the coil in the magnetic field, it causes varying movement of the coil along the axis
with the frequency of the current variations. The diaphragm is thus set into vibrations
which are communicated to the external air and the sound is reproduced. The greater
the energy supplied to the voice coil, the louder will be the sound emitted by the
diaphragm. Completely surrounding the cone and attached to it by silk threads is the
'baffle'. It prevents the air vibrating behind the cone from flowing round to its front.
The relation between the current and driving force is linear and force is
independent of the position of the coil in the gap for considerable movements. When
suitably designed a fairly uniform response of 80cps to 1000cps is secured. It is
capable of radiating large power without appreciable asymmetric distortion.

ACOUSTICAL MEASUREMENTS:
The measurement of airborne and waterborne sound is of increasing interest
to engineers. Airborne sound measurements are important in the development of less
noisy machinery and equipment, in diagnosis of vibration problems, and in the
design and test of sound recording and reproducing equipment. In large rocket and
jet engines, the sound pressures produced by the exhaust may be large enough to
cause fatigue failure of metal panels because of vibration. Waterborne sound has
been applied in underwater direction and range finding equipment like "sonar". Most
sound transducers are basically pressure measuring devices.
The basic definitions of sound are in terms of the magnitude of fluctuating
component of pressure in a fluid medium. The sound pressure level (SPL) is defined
by SPL = Sound pressure level =
P
.0002
10
20Log decibels (dB) [1]
p - root mean square (rms) sound pressure,

bar.
The rms value of fluctuating component of pressure is used in equation I,
because most sounds are random signals rather than pure sine waves. The value
0.0002 bar is an accepted standard reference value of pressure against which other
pressure s are compared by equation1. When p = 0.0002 bar, the sound pressure
level is 0dB
SOUND LEVEL METER
The most commonly used instrument for sound measurement is the sound
level meter. This actually made up of a number of interconnected components. The
sound pressure is transduced to a voltage by means of a microphone. Microphones
generally employ a thin diaphragm to convert pressure to motion. Microphones often
have an arrangement so that it will not respond to constant and slowly varying
responses. This is necessary because the hour-to-hour and day-to-day changes in
atmospheric pressure are much greater than the sound pressure fluctuations to which
the microphone must respond. The motion is then converted to voltage by some
suitable transducer, usually a capacitance, piezo-electric or moving coil type.
The output voltage of the microphone generally is quite small and at a high
impedance level. So an amplifier of high input impedance and gain is used at the
output of the microphone. This can be a relatively simple ac amplifier, since response
to static or slowly varying voltages is not required.
Following the first amplifier are the weighting networks. They are electrical
filters whose frequency response is tailored to approximate the frequency response of
the average human ear. Readings taken with a weighting network are called sound
level rather than sound pressure level.
The output of the weighting network is further amplified and an output jack
provided to lead this signal to an oscilloscope or to a spectrum analyzer. If only the
overall sound magnitude is desired, the rms value of e
3
must be found. The average
value of e
3
is determined by rectifying and filtering and then the meter scale is
calibrated to read rms values. This procedure is exact for pure sine waves since there
is a precise relation between the average value and rms value of a sine wave. For non
sinusoidal wave this is not true, but the error is generally small enough to be
acceptable for relatively unsophisticated work.
ACOUSTICS OF BUILDINGS
Factors affecting the acoustics of a building are: Reverberation, Loudness,
Focusing, Echelon effect, extraneous noise, Resonance.
Reverberation
It is observed that for a listener in room or auditorium, whenever a sound
pulse is produced, he receives directly sound waves from the source, as well as sound
waves from the walls, ceiling and other materials present in the room. The waves
received by the listener are:
1. direct waves
2. Reflected waves due to multiple reflections at the various surfaces.
The quality of the note received by the listener will be the combined effect
of these two sets of waves. There is also a time gap between the direct wave received
by the listener and the direct waves received by the listener and the waves received
by successive reflection. Due to this, the sound persists for sometime even after the
source has stopped. This persistence of sound is termed as reverberation. The time
gap between the initial direct note and the reflected note up to the minimum
audibility level is called reverberation time. The reverberation time will depend on
the size of the room or the auditorium, the nature of the reflecting material on the
wall and the ceiling and the area of the reflecting surfaces.
In a good auditorium it is necessary to keep the reverberation time
negligibly small. The intensity of sound received by the listener is shown graphically
in fig. When a source emits sound, the waves spread out and the listener is aware of
the commencement of the sound when the direct waves reach his ears. Subsequently
the listener receives sound energy due to reflected waves also. If the note is
continuously sounded, the intensity of sound at the listener's ear gradually increases.
After sometime a balance is reached between the energy emitted per second by the
source and the energy lost or dissipated by walls or other materials. The resultant
energy attains an average steady value, and to the listener the intensity sound appears
to be steady and constant. When the intensity of sound falls below the minimum
audibility level, the listener will not hear the sound.
When a series of notes are produced in the auditorium (say speech or music)
each note will give raise its own in intensity curve with respect to time. For clear
audibility of speech or music, it is necessary that (1) each separate note should give
sufficient intensity of sound in every part of the auditorium, and (2) each note should
die down rapidly before the maximum average intensity due to the next note is heard
by the listener as in fig. This is particularly important with speech. In case of music
comparatively more reverberation can be tolerated.
Loudness:
The speech of a person in a hall can be heard by an audience consisting of
about 100 persons. However, to ensure uniform distribution of sound intensity in the
hall, electrically amplified loud speakers are used. These speakers are kept at
different places in the auditorium and are located generally at a height higher than
the speaker's head. Amplifiers however make the low frequency tones more
prominent and hence the amplification has to be kept low. The presence of low
artificial ceilings improves the audibility in general.
Focusing:
The presence of cylindrical or spherical surfaces on the walls or the ceiling
gives rise to undesirable focusing. In fig. the observer also receives the sound waves
after reflection from the ceiling. Thus the intensity of sound at O is comparatively
higher than other positions in the auditorium. It may also happen that the direct and
the reflected waves are in the opposite phase. This results in minimum intensity of
sound at O. Further the direct and the reflected waves may form a stationary wave
pattern. This causes distribution of sound intensity.
Echelon Effect
If there is regular structure similar to a flight of stairs or a set of railings in
the hall, the sound produced in front of such a structure may produce a musical note
due to regular successive echoes of sound reaching the observer. Such an effect is
called Echelon effect. If the frequency of this note is within the audible range, the
stair cases are covered with carpets to avoid reflection of sound.

Extraneous Noise
The extraneous noise may be due to (1) sound received from outside the
room (2) the sound produced by fans etc., inside the auditorium. The external sound
cannot be completely eliminated but can be minimized by using double or triple
windows and doors. Proper attention must also be paid to maximum permissible
speed of fans and rate of air circulation in the room. The air conditioning pipes
should be covered with cork and insulated acoustically from the main building.
Resonance
The acoustics of a building may also be affected by resonance. If there is
resonance for any audio frequency note, the intensity of the note will be entirely
different from the intensity desired. In halls of large size, the resonance frequency is
much below the audible limit and harmful effects due to resonance will not be
present.
Sound Distribution in an Auditorium
The design of an auditorium requires smooth decay and growth of sound. In
an auditorium, the sound must be distributed or diffused over the whole area. To
ensure these factors, acoustic treatment is given viz., scattering effect of objects,
irregularities on the wall surfaces, fixing absorptive material on the walls etc.
The first reflection of sound waves at different positions of ceiling. It is
clear from the figure that the reflected sound is distributed evenly in the auditorium
viz., the main floor and the balcony. This design enables an even distribution of
sound intensity.
Requisites for Good Acoustics
The reverberation of sound in an auditorium is due to multiple reflections
taking place at various surfaces present within the auditorium. The acoustics of an
auditorium can be improved by using the surfaces with high absorption coefficient.
This will reduce the reverberation time below the optimum value. This can be
achieved as follows.

1) By hanging Heavy Curtains.
2) By hanging Picture and Maps.
3) By having a few Open Windows.
4) By having Good Audience. Each person is equivalent to about 0.50m
2

area of an open window.
5) The curved walls and comers bounded by two walls should be avoided.
This is done to avoid 1. Concentration of sound 2. Dead spaces.
6) Upholstered seats should be provided so that the absorption is
approximately the same with or without the audience.
7) The walls and the ceiling should be covered with the materials having
high absorption coefficient i.e., with perforated cardboards, felt asbestos fibre
glass etc.
8) The walls should be engraved and made rough with decorative
materials to increase absorption.
FOURIER THEOREM
Fourier theorem deals with the summation of a number of simple harmonic
vibrations, in which the vibrations are in the same straight line. The theorem also
helps in the synthesis and analysis of complex forms of vibrations. This theorem was
formulated by J.B.T. Fourier in 1828.
The theorem states that "any single valued periodic function can be
expressed as a sum of a number of simple harmonic terms which are multiples of the
given function." The theorem is generally referred in relation to the study of
transverse vibration of strings. However, the theorem has a wider scope.
The theorem is valid only if the following conditions are satisfied.
1. The displacement must be single valued function and continuous. This
condition is satisfied in all cases of mechanical vibrations because a
single particle cannot actually have two different displacements
simultaneously.

2. The displacement must always have a finite value. This is true in the
case of sound.
Fourier Series
The Fourier series can be expressed by the series.
0 1 1
y f ( t) A A cos( t ) = e = + e +o =
2 2 m m
A cos(2 t ) ......... A cos(m t ) e +o + + e +o
This can be written in the form
m
m m
m 1
y f ( t) A cos(m t )
=
=
= e = e + c

Here y represents the displacement of the complex periodic vibrations of
frequency
2
e | |
|
t
\ .
. A
1
, A
2
... A
m
are amplitudes of the components of the 4 simple
harmonic vibrations
1 2 m,........,
, ,........, o o o represent their respective initial phases.
It may be mentioned that sometimes it is convenient to represent y as a sum
of sine and cosine series in the form.
| |
m
m m
m 0
y f ( t) A sinm t B cosm t)
=
=
= e = e + e

Here A
0
= 0
m m
0 m m
m 1 m 1
y f ( t) B A sinm t B cosm t
= =
= =
= e = + e + e


The method of finding the amplitude of Fourier coefficients (B
0
, A
m
and B
m

for all values of m is called Fourier analysis.
1
0
0
l
B ydt
T
=
}

1
m
0
2
A ysin(m t)dt
T
= e
}

1
m
0
2
B ycos(m t)dt
T
= e
}

Where T is time period of the function
Fourier series has been employed for the analysis of complex periodic
curves with a view to finding the various harmonic components of which they may
be built up together with their amplitudes and relative phases. It can be shown in
various ways that such components have an objective physical existence and are not
a mere mathematical fiction. A trained musical ear can easily resolve complex
musical sound hearing each simple harmonic component as a separate simple tone
and thus serve as a natural Fourier analyzer.
For practical purposes the-waveforms of different sounds are recorded and
analyzed by various methods. The coordinates of the complex curve under study at
different instants are determined. Then substitute the data obtained in the above
equations for obtaining the Fourier coefficients. Thus the various harmonic
components and their relative amplitudes are ascertained.
ACOUSTICAL IMPEDANCE AND FILTERS
An acoustic filter is a device which has been extensively used for analyzing
the quality of complex sound waves. It is so designed that it transmits certain
selected ranges of frequency with negligible attenuation, and suppresses other
frequencies almost entirely. The analogy between acoustic filters and electrical filters
used in ac circuits for a similar purpose is so close that the considerations and
equations operating in the functioning of the latter have helped a good deal in
designing the former.
If we consider the equation of motion of a body executing forced vibration
in a resisting medium
2
2
d y dy
m ky Fsin t
dt dt
+ + = e ................ 1
The corresponding equation in electricity is
2
2
d q dq q
L R Esin t
dt c dt
+ + = e ................ 2
The following are mechanical analogues of the corresponding electrical
quantities.
Mechanical Electrical
Force (F) Emf (E
0
)
Mass (m) Inductance (L)
Resistance ( ) Resistance
Stiffness (k) l/Capacitance
Displacement (y) Charge (q)
Velocity (dy/dt) Current (dq/dt)
Current does not flow along a single line but is branched at several points and so in
the case of acoustic filters, where the main acoustic tube or conductor is provided
with several branch tubes of suitable dimensions to serve as guide for the waves in
the same way as electrical transmission line with its several branches guides the
current. Several other conditions must be satisfied before the analogy is complete in
every respect.
1. The length of
any selected section of an acoustic transmission line must be small as
compared to the wave length so that no change of the phase occurs within it.
2. The algebraic
sum of the volume displacements at any junction of the line is zero just as in
Kirchhoffs law.
Corresponding to the electrical impedance there is the acoustical impedance
which is defined as the ratio of the applied pressure difference and the rate of change
of volume displacement.
There are two types of acoustic filters:
1. Low pass Filters:
These are made up of two concentric cylinders joined by walls equally spaced
and perpendicular to the axes. Each chamber thus formed had row apertures in
the inner cylinder which served as the transmission tube.
2. High pass filter:
These are made with a straight tube for transmission and short side tubes
0.5cm long and 0.28cm diameter opening through a hole with conductivity
0.08 into a tube 10cm long and 1 cm diameter. Six sections of such a filter
would transmit about 90 percent of sounds above 800 Hz but would refuse
transmission to sounds of lower frequency.