AD Unit-4

AD: Audiometers & Hearing Aids - 1
AUDIOMETRS AND HEARING AIDS

MECHANISM OF HEARING
 Sound waves are longitudinal waves in which the motion of each particle of the medium in
which the wave is travelling, moves backward and forward along a line in the direction in
which the wave is propagated. The human aural system reacts to these oscillating pressure
changes and transmits them to the brain through a series of steps.
 Every sound produces sound waves or vibrations in the air, which travel at about 332 metres
per second.
 The frequency of a sound vibration is its pitch. The higher the frequency of vibration, the
higher is the pitch. The larger the intensity (size or amplitude) of the vibration, the louder is
the sound.
 The sounds heard most acutely by the human ear are those from sources that vibrate at
frequencies between 500 and 5000 hertz. The entire audible range extends from 20 to
20,000 Hz. Sounds of speech primarily contain frequencies between 100 and 3000 Hz.
 The following events are involved in hearing,

1. The auricle directs sound waves into the external auditory canal toward the tympanic
membrane. When sound waves strike the tympanic membrane, the alternating waves of
high and low pressure in the air cause the tympanic membrane to vibrate back and forth.
The tympanic membrane vibrates slowly in response to low-frequency (low-pitched) sounds
and rapidly in response to high frequency (high-pitched) sounds.
2. The central area of the tympanic membrane connects to the malleus, which vibrates along
with the tympanic membrane. This vibration is transmitted from the malleus to the incus
and then to the stapes.
3. The stapes moves back and forth and vibrates the oval window. The vibrations at the
oval window are about 20 times more vigorous than the tympanic membrane because the
auditory ossicles efficiently transmit small vibrations spread over a large surface area (the
tympanic membrane) into larger vibrations at a smaller surface (the oval window).
4. The movement of the stapes at the oval window sets up fluid pressure waves in the
perilymph of the cochlea. As the oval window bulges inward, it pushes on the perilymph of
the scala vestibuli.
5. The pressure waves travel through the perilymph of the scala vestibuli, then the vestibular
membrane, and then move into the endolymph inside the cochlear duct.
6. The pressure waves in the endolymph cause the basilar membrane to vibrate, which moves
the hair cells of the spiral organ against the tectorial membrane. This leads to bending of the
stereocilia and ultimately to the generation of nerve impulses in first-order neurons in
cochlear nerve fibers.
Sound waves of various frequencies cause certain regions of the basilar membrane to vibrate
more intensely than other regions. Each segment of the basilar membrane is “tuned” for a
particular pitch.
Because the membrane is narrower and stiffer at the base of the cochlea (closer to the oval
window), high-frequency (high-pitched) sounds induce maximal vibrations in this region.
Toward the apex of the cochlea, the basilar membrane is wider and more flexible; low-
frequency (low-pitched) sounds cause maximal vibration of the basilar membrane there.
Loudness is determined by the intensity of sound waves. High-intensity sound waves cause
larger vibrations of the basilar membrane, which leads to a higher frequency of nerve
impulses reaching the brain. Louder sounds also may stimulate a larger number of hair cells.
7. Pressure waves are transmitted from the scala vestibuli to the scala tympani and eventually
to the round window, causing it to bulge outward into the middle ear.
Sound transduction
 During the process of hearing, the hair cells convert mechanical vibrations of sound waves
into action potentials in auditory nerve fibers. This process is called sound transduction.
 Hair cells are epithelial, but function somewhat like neurons. For example, when a hair cell
is at rest, its membrane is polarized. When it is stimulated, selective K channels open,
depolarizing the membrane and making it more permeable to calcium ions.
 The hair cell has no axon or dendrites, but it has neurotransmitter-containing vesicles near
its base. As calcium ions diffuse into the cell, some of these vesicles fuse with the cell
membrane and release a neurotransmitter.
 The neurotransmitter stimulates the ends of nearby sensory neurons, and in response they
transmit action potentials along the cochlear branch of the vestibulocochlear nerve to the
auditory cortex of the temporal lobe of the brain.
Auditory Pathway
 Sensory nerve fibers originating in the spiral organ lead to the cell bodies of bipolar neurons
in the spiral ganglion, and then continue as fibers of the cochlear nerve. This nerve joins the
vestibular nerve and the two together become the vestibulocochlear nerve (cranial nerve
VIII).
 The vestibulocochlear nerve exits the internal acoustic meatus of the temporal bone, thus
leaving the inner ear and entering the cranial cavity. It ends a short distance later at the
medulla oblongata.
 Cochlear nerve fibers project to the cochlear nucleus on each side of the medulla.
 Once they leave the cochlear nucleus, most of the axons of the cochlear nucleus cells cross
over to the opposite side (contralateral side) of the brain.
 Both crossed and uncrossed fibres from the cochlear nuclei synapse in the area of the
brainstem called the superior olivary complex. This is the first place in the ascending
pathway to receive information from both ears.
 Neural impulses are transmitted from the superior olivary complex to the inferior colliculus
through and/or around the lateral lemniscus (some fibres synapse in the lateral lemniscus
but most travel through it to the inferior colliculus), from there to the medial geniculate
body and finally to the the primary auditory area of the cerebral cortex in the temporal lobe
of the cerebrum.
Auditory Pathway
 Another system follows a similar path, but in reverse, from the cortex to the cochlear nuclei.
This is the descending auditory pathway. In general, the descending pathway may be
regarded as exercising an inhibitory function by means of a sort of negative feedback. It may
also determine which ascending impulses are to be blocked and which are allowed to pass
to other centres in the brain. The olivocochlear bundle, which arises from the olivary
complex, is involved in sharpening or otherwise modifying the analysis that is made in the
cochlea.
Air and Bone Conduction
 Air conduction, by definition, is the transmission of sound through the external and middle
ear to the internal ear.
 Bone conduction, on the other hand, refers to transmission of sound to the internal ear
mediated by mechanical vibration of the cranial bones and soft tissues.
HEARING LOSS
 The two most common forms of hearing loss are
 Conductive loss, which refers to dysfunction of the outer or middle ear.
The outer and middle ears are largely linear systems, and consequently, conductive
dysfunction typically produces a simple linear reduction in the amplitude of acoustic signals
as they are transmitted to the inner ear. Thus, loud sounds become quieter and quiet sounds
may become inaudible.
External ear and outer meatus may cause conductive losses by malformations, stenoses, ear
wax (cerumen), proliferations of bony tissue around the stapes footplate, debris, or foreign
objects.
Conductive losses can never cause complete deafness because the inner ear is excited by
bone conduction even in the case of a total disruption of the middle ear apparatus. The
maximum conductive hearing loss is around 50 dB.
 Sensorineural loss, in which the inner ear or the auditory pathways of the brain are impaired.
The effects of sensorineural loss are more multifaceted. Because the cochlea is a highly
nonlinear system, any impairment of cochlear structures (particularly the inner and outer
hair cells) can lead to substantial distortion in the neural representation of acoustic signals,
in addition to loss of audibility.
Common causes of sensorineural impairment include exposure to loud sounds, aging,
disease, head injury, and ototoxic drugs.
Many of these inner ear diseases, the degree of hearing loss can be absolute (complete
deafness).
Measurement of sound
AUDIOMETRY
 Audiometry (from Latin: audīre, "to hear" and metria, “to measure") is the testing of hearing
ability, involving thresholds and differing frequencies.
 Typically, audiometric tests determine a subject's hearing levels with the help of an
audiometer, but may also measure ability to discriminate between different sound
intensities, recognize pitch, or distinguished speech from background noise. Acoustic reflex
and otoacoustic emissions may also be measured.
 Results of audiometric tests are used to diagnose hearing loss or diseases of the ear, and
often make use of an Audiogram.
AUDIOMETER
 An audiometer is a device used for evaluating and quantifying hearing loss of an individual.
It measures the hearing sensitivity by determining the individual's hearing threshold for pure
tones and speech.
Components of audiometer
 Regardless of audiometer type, there are three main components to any audiometer, as
shown schematically in Figure.
 The primary components are

1. Oscillator,
2. Amplifier,
3. Attenuator,
4. Interrupter switch and
5. Transducer.
 The oscillator generates pure tones, usually at discrete frequencies at the octave and mid-
octave frequencies.
An octave is a doubling of frequency. Most audiometers produce octave intervals from 125
Hz to 8 kHz.
The frequencies generated are 125, 250, 500,750, 1000, 1500, 2000, 3000, 4000, 6000 and
8000 Hz.
 Amplifier amplifies the produced oscillations to a fixed intensity level (e.g.110 dB HL). The
most important characteristic of the amplifier is that it produces very little distortion, and
has a good signal to noise ratio. In most audiometers the power amplifier is run at constant
high signal output levels.
 The attenuator controls the level of the signal from the audiometer. The tones are
attenuated with the use of a manual dial or electronic attenuator, which is numbered
(contrary to attenuation) in decibels above the normal threshold for each frequency.
The sound level can also be varied, usually in 5 dB steps, from –10 dBHL to 110 dBHL or more.
 The interrupter switch controls the duration of the signal that is presented to the patient.
The interrupter switch is typically set to the off position for pure-tone signals and is turned
on when the presentation button is pressed. The interrupter switch is typically set to the on
position for speech signals.
The tones presented to the patient should be switched on and off. This feature is important
because a continuous tone undergoes decay during a period of time. This switch gives the
option of providing the tone in a continuous or an interrupted manner.
 Transducers are the devices that convert the electrical energy from the audiometer into
acoustical or vibratory energy. Transducers used for audiometric purposes are
Ear phones: They are usually of the moving coil type. They give a reasonably flat frequency
response up to 6 kHz after which their sensitivity falls rapidly. They are not speciality
designed for audiometric applications but for communication purposes. In their miniature
form, they are used in hearing aids. When used via insert earphone and ear moulds, they
provide greater acoustic power to be transferred to the small volume of the external ear. It
may be noted that audiometer earphones are not interchangeable and must remain
identified with a specific instrument to preserve its calibration.
In conditions of ambient noise being too high for unshielded earphones, specially designed
audio caps are used. They use a fully-articulated suspension system which leaves the
standard ear caps free to locate against the pinnate with normal pressure, and at the same
time to enclose fully the external ears with noise-excluding shells, sealed with soft plastic
cushions, to exclude background noise which will otherwise result in elevated threshold
measurements.
Microphones: These are used to translate wave motion in air into electrical signal.
Usual types are:
(i) Carbon button which changes resistance with air pressure,
(ii) Electrodynamic where a voltage is induced in a coil by its motion relative to a magnet,
(iii) Condenser where capacitance of a condenser is varied by the vibration of one of the
condenser plates. High quality condenser microphones of diameters 12.5, 6.25 and 3.125
mm are currently used, depending on the frequency- to be- measured.
For special purposes, microphones can be fitted to the ear caps and used in reciprocal
arrangement to transmit sound to the ear.
Bone Vibrators: In the early days, bone vibrators were composed of a rod connected to an
electromagnetically excited driving system. They were often held by hand against the
mastoid process. Such units were large and cumbersome and are not in clinical use today. In
the present form, bone vibrators are of the hearing-aid type in which the transduction
mechanism changes the alternating current into a vibrator force through a diaphragm. The
diaphragm and its basic mechanical parameters like mass, compliance and resistance are
important in establishing is response characteristics. Though convenient, it is a very
inefficient means of transduction and has a rather limited and peaky frequency response.
The plane circular contact area of a bone vibrator is recommended to be 175 ± 25 mm 2. It is
held in position by a headband.
Loudspeakers: They are used to deliver auditory stimuli, when it is not possible to have close
coupling of the transducer to the ear. Loud speakers are used in testing paediatric patients.
Obviously, the acoustic energy loss into the surroundings will be much greater than when
stimulation is applied directly via in earphone. The acoustics of the test room and masking
of the non-test ear are two important factors which merit special considerations while using
the sound field of a loudspeaker.
Types of Audiometry
1. Pure tone audiometry
2. Speech audiometry
1. PURE TONE AUDIOMETRY
 Pure tone audiometry measures hearing sensitivity for a series of single frequency sounds
(pure tone) within the range of normal hearing.
 A wave in air, which involves only one frequency of vibration, is known as pure-tone.
 Pure tone audiometry (PTA) is the routine hearing test used to identify hearing threshold
levels of an individual, enabling determination of the degree, type and configuration of a
hearing loss.
Instruments
 The pure-tone audiometer is an instrument which produces sounds, in the form of pure
tones, which can be varied both in frequency and intensity.
 A pure-tone audiometer basically consists of an LC oscillator in which the inductance and
tuning capacitance are of close tolerances for having a precise control on the frequency of
oscillations. The oscillator is coupled to an output current amplifier stage to produce the
required power levels. The attenuators used in these instruments are of the ladder type, of
nominal 10 Ω impedance. The signals are presented acoustically to the ear by an earphone
or small loudspeaker.
 The audiometer contains an oscillator which produces a sinusoidal waveform. The frequency
of this sine wave can be changed, and the minimum available frequencies are 250, 500, 1000,
2000, 4000 and 8000 Hz.
 The output from the oscillator is taken to an audio amplifier and then into an attenuator,
which may be either stepped or continuously variable. A standard range would be from −10
to 120 dB.
 The output from the attenuator is taken to the headphones.
 Pure tones are presented to the patient either through headphones for air conduction
measurements, or through a bone conductor for bone conduction measurements.
Pure tone tests
 A pure-tone audiometer is used primarily to obtain air-conduction and bone-conduction
thresholds of hearing. These thresholds are helpful in the diagnosis of hearing loss.
Air conduction audiometry
 In conventional pure-tone audiometry, head phones are worn by the subject and a set of
responses is obtained for air-conducted sounds directed to each ear in turn.
 When hearing is measured with pure tones presented through headphones, this
measurement is called air conduction. The sounds go via the air, down the ear canal, through
the middle ear, and to the very delicate organ of hearing in the inner ear—the cochlea.
 The patient listen pure tones and indicate when they can hear them, generally by pressing
a button.
 The loudness of each tone is reduced until you can just hear the tone. The softest sounds
the patient can hear are known as hearing thresholds, and these are marked on a graph
called an audiogram.
 To avoid ambient noise affecting the measurements, testing is Often conducted inside a
sound booth.
Procedure
 Place the headphones comfortably on the patient, making sure that the red phone is
over the right ear. Spectacles can be most uncomfortable when headphones are worn
and are therefore best removed.
 Start at a level of 50 dB and 1000 Hz. The reason that the test commences with 1000 Hz
is
because this falls in the middle of the most sensitive area of the hearing spectrum. It is
also a clear tone to hear for a person who has never been tested before
 Present tones of about 2 s in duration with varying intervals (1–3 s).
 If the tone is heard, then reduce the level in 10 dB steps until it is no longer heard. If the
starting tone is not heard, then raise the level in 20 dB steps until it is heard, and then
descend in 10 dB steps.
 After testing at 1000 Hz proceed to 2000, 4000 and 8000 Hz. Repeat the reading at 1000
Hz and then make measurements at 500, 250 and 125 Hz.
 Great care must be taken to vary the interval between the tones in order to detect where
incorrect responses are given.
Bone conduction audiometry
 The sensitivity of the cochlea can also be tested by placing a small vibrator on the mastoid
bone behind the ear and again measuring the softest sounds that can be heard. The vibrator
is usually attached by a sprung band passing over the head.
 Sounds presented this way travel through the bones of the skull to the cochlea and hearing
nerves, bypassing the middle ear. This type of testing is called bone conduction.
 The air conduction and bone conduction hearing levels on the audiogram can reveal where
hearing loss is conductive, mixed or sensorineural.
1. If the air conduction thresholds show a hearing loss but the bone conduction thresholds
are normal, indicate hearing loss can be attributed to conductive hearing loss. Reasons
could include fluid in the middle ear, excessive wax, a perforated tympanic membrane
or that the bones of the middle ear are not functioning normally.
2. Normal bone conduction with abnormal air conduction indicates a middle ear pathology.
3. When the hearing threshold in both tests is more or less equal, it is likely to be a
sensorineural problem.
 Bone conduction testing is similar to the procedure for testing air conduction.
2. SPEECH AUDIOMETRY
 Speech audiometry measures hearing sensitivity and speech discrimination in conversation.
Speech audiometry is used to measure the ability of a patient to perceive speech signals.
 Speech audiometry assesses a patient’s auditory ability using words, which are much more
representative of everyday listening experience than pure tones.
 Measuring the ability to perceive speech gives the clinician a clearer picture of the patient’s
functional hearing ability and is extremely valuable to predict a patient’s success with
hearing aids.
Speech Audiometer
 Speech audiometry is conducted in the ‘‘speech mode’’ setting of a clinical audiometer.
 Specially designed speech audiometers are used for this purpose. They incorporate a good
quality tape recorder, which can play recorded speech. A double band tape recorder is
preferred to interface the two channel audiometer units. Masking noise is supplied by the
noise generator. The two channels supply the two head-phones or the two loud speakers
which are of 25 W each.
 The tape recorder has a capacity for recording a limitless variety of test material and a
consistency of speech input, which cannot be obtained for live-voice audiometry in relation
to test-retest repeatability. Another advantage of the tape recorded material is that the test
words and sentences can be selected to cater for the widely differing needs of age,
intelligence, dialect and language.
 In speech audiometers, live-voice facilities are incorporated primarily for communication
purposes as the inherent unreliability of live-voice speech tests may lead to serious errors.
The microphone amplifier used for this purpose is a simple two stage amplifier.
 The frequency response characteristics of a live-voice channel should be such that with the
microphone in a free sound field having a constant sound pressure level, the sound pressure
level developed by the earphone of the audiometer in the artificial ear at frequencies in the
range 250 to 4000 Hz does not differ from that at 1000 Hz by more than 110 dB. Also, it shall
not rise at any frequency outside this band by more than 15 dB, relative to the level at 1000
Hz.
 Speech stimuli are presented through the same types of transducers as those used for pure
tone audiometry.
 Recorded speech materials typically include a calibration tone, and the input level is adjusted
for individual recordings to a specified intensity level.
Speech-Threshold Testing
 In general, speech audiometry is conducted with the examiner in one room and the listener
in another. With this arrangement, the examiner is able to observe the listener and maintain
easy communication through microphones in both rooms, but the speech stimuli can be
presented under carefully controlled conditions.
 Speech thresholds may be of two kinds:
1. The speech-detection threshold (SDT) and
2. The speech-recognition threshold (SRT).
1. Speech Recognition threshold (SRT)
 The speech recognition threshold (SRT) is the lowest intensity level at which the patient
correctly responds to (repeats, writes down, points to) approximately 50% of the words.
 This level should correspond roughly to the average of the pure tone audiometry thresholds
at 500, 1000 and 2000 Hz.
 Speech recognition threshold (SRT) testing typically entails presentation of spondees (two-
syllable, compound words), spoken with equal stress on each syllable (e.g., baseball,
toothbrush, airplane).
 The SRT is tested one ear at a time, beginning with the better ear first. The patient is
instructed to repeat the words they hear, even if the words are very faint.
 The words are then presented to the patient at a comfortably loud level in order to
familiarize the listener with approximately 8 to 10 test word items that will be used for the
test.
 The clinician proceeds to search for the threshold by adjusting the volume of these Same
words (lowering by 10 dB each time the patient correctly identifies a word, and raising by 5
dB each time the response is incorrect), much in the same way that pure tone air conduction
thresholds are tested.
 The softest decibel level at which the listener is able to correctly identify 50% of these simple
words is the SRT for that ear.
 This procedure is then repeated for the other ear, and the results are reported accordingly.
2. Speech-Detection Threshold
 The speech-detection threshold (SDT) may be defined as the lowest level, in decibels, at
which a subject can barely detect the presence of speech and identify it as speech. The SDT
is sometimes called the speech-awareness threshold (SAT).
 Speech discrimination can be measured using a variety of test materials. Monosyllabic single
words preceded by a carrier phrase are most frequently used.
 The SDT is tested one ear at a time, beginning with the better ear first using traditional or
insert earphones.
 If thresholds for spondaic words cannot be established, because of language impairment or
other limitations such as young age or inability to speak because of injury, the SDT may
represent a useful estimate of the level at which the patient indicates awareness of the
presence of speech.
 In this type of speech threshold testing, the patient is not required to repeat the speech
stimulus, which may be just a simple word or nonsense sound, but, instead, the patient
simply responds with a hand movement or other gesture to indicate that a sound was
detected.
 If audiometric results (air and bone conduction pure tone audiometry) indicate normal
hearing or a conductive loss, start testing at 30dB above the measured SRT, or if the SRT is
not available, 30dB above the average pure tone thresholds measured at 500, 1000 and 2000
Hz.
 The intensity is then lowered by 10 dB each time the patient detects the sound, and raised
by 5 dB each time the sound has not been detected. Just as in pure tone air conduction
testing, the intensity level of the sound.
 Patients may respond verbally, with hand or finger signals, or with a push-button, indicating
the lowest level, in dB HL, at which they can barely detect speech.
MASKING IN AUDIOMETRY
 When sound is applied to one ear, its vibration is conducted through the bone of the skull to
the contralateral cochlea. When the stimuli presented to the test ear stimulates the cochlea
of the non-test ear, this is known as cross hearing.
 Masking is a technique to remove the effect of cross hearing during testing by temporarily
presenting noise at a predetermined level.
Effect of Masking
 A hearing loss which only affects one ear is called unilateral and a loss to both ears, but of
different degrees, is called asymmetrical.
 If a patient has a much greater hearing loss in one ear than the other then it is possible that
sounds presented to the poor ear may be heard in the good ear.
 The masking noise temporarily elevates the threshold of the non-test ear, thereby
preventing the non-test ear from detecting the test signal presented to the test ear.
Therefore, hearing thresholds obtained with masking provide an accurate representation of
the true hearing threshold level of the test ear.
Masking concepts
 Test ear (TE): Ear of which air or bone conduction threshold is being measured
 Non-test ear (NTE): Opposite or contralateral ear
 Interaural attenuation (IAA): A reduction or loss of energy occurs with cross hearing, which
is referred to as interaural attenuation (IA) or transcranial transmission loss. It is one of the
most important concepts related to masking.
Minimum levels of IAA guide the tester as to when crossover is likely to occur. The values
differ for air and bone conduction. IAA for bone conduction is approximately 0 dB. Because
IAA for air conduction (AC) is about 40dB, there is no risk of crossover when testing air
conduction if the difference between the test ear and the nontest ear bone is <40 dB.
Interaural attenuation with air conduction. Interaural attenuation with bone

conduction
IAA also depends on the type of transducer being used; levels are set as a function of the
transducer type:
Supra-aural earphones = 40 dB;
Insert ear-phones = 50 - 70 dB;
BC vibrator = 0 dB
Masking criteria
 The criteria to assess when masking is needed, are:
 If the both ears are normal, masking is not needed.
 If the difference between left and right unmasked air conduction thresholds is 40 dB or
more, one should always mask air conduction.
 For bone conduction testing, the sound always goes to both ears, and one cannot test
one ear by itself (usually) without using masking. The best strategy therefore, for bone
conduction testing, is to always mask.
Types of masking noise
 Several different kinds of masking noises are available on commercial pure-tone

audiometers. Each noise has a characteristic spectrum and therefore provides a different
degree of masking efficiency at different frequencies.
 There are three types of masking noise:

1. White noise consists of a wide range of frequencies with uniform loudness and is used
to mask tones or speeches.
2. Pink noise is similar to white noise except that it consists of a higher proportion of lower
frequencies and is used to mask speech alone.
3. Narrow-band noise consists only of frequencies close to the ones being tested and is
used to mask pure tones.
 Masking may also be required when carrying out speech audiometry. Speech-shaped or
speech equivalent noise is normally used for this purpose. This type of noise is similar to pink
noise in that it has its greatest intensity in the low frequencies.
Procedure for masking
 The thresholds for the masking noise must be determined so that the level of masking
applied is appropriate.
 Too little masking will not be effective. If too much masking is applied cross-masking
might occur, where the masking noise is heard in the test ear and interferes with the
pure-tone threshold determination.
 Although it is important to be accurate, extreme precision in determining the threshold
for masking is not required. It is not necessary to perform a test such as that used for
finding the pure-tone threshold in order to establish a masking noise threshold. The
threshold of masking is denoted M, in the British Society of Audiology’s

recommendations.
 Measure the threshold of the masking signal in the non-test ear.
 Present the tone to the poor ear at unmasked threshold level.
 Introduce narrow-band masking into the good ear at the masking signal threshold.
 Now present the tone to the poor ear again:
1. If the patient still hears the tone then increase the masking level to the good ear in 5
dB steps up to a maximum of 30 dB above threshold. If the tone is still heard then
this is considered to be the true threshold for the poor ear.
2. If the patient does not hear the tone then increase the intensity of the tone presented
to the poor ear in 5 dB steps until it is heard. Then proceed as in 1.
3. The test is not considered satisfactory until the tone in the poor ear can be heard for
an increase of 30 dB in the masking to the good ear.
THE AUDIOGRAM
 The results of basic audiometry may be displayed in numeric form or on a graph called an
audiogram. The audiogram is a graphical display of the hearing test.
 Frequency: Frequency or pitch is measured in Hertz (Hz). The higher the number, the higher
the pitch of the sound. Frequencies range from low-pitch to high-pitch and read from left to
right on the audiogram. Each vertical line represents a different frequency.
Although the normal human ear can detect frequencies below 100 Hz and as high as 20,000
Hz, the audible frequency range most important for human communication lies between 125
and 8000 Hz, and the audiogram usually depicts this more restricted range. The ones used
most often during testing are 250, 500, 1000, 2000, 4000 and 8000 Hz.
 Intensity: The intensity is measured in decibels (dB). The intensity relates to how loud or soft
a sound is. Each horizontal line represents a different intensity level. The softest sounds are
at the top of the chart and the loudest sounds at the bottom.
The softest intensity tested is typically 0 dB and the loudest is 120 dB. Zero decibels (0 dB)
does not mean ‘no sound’. It is just extremely soft. Conversational voice level is around 65
db.
 Speech Testing: Speech discrimination or word recognition ability is scored as a percentage.

This score represents how well a list of words could be repeated. The words are presented
via headphones at a comfortable volume level with no background noise present.
Standard Symbols
 Audiogram provides a separate graph for each ear, although in some cases the results from
both left and right ears are plotted on one graph.
 It is important to use standard symbols on an audiogram to avoid confusion.
 The symbols are shown in red for the right ear and blue for the left ear.
 The air conduction (AC) symbol for the right ear is a circle, O (unmasked) or a triangle Δ
(masked), preferably drawn in red, and the standard symbol for the left ear the symbol is a
cross, X (unmasked) or a square, □ (masked). Some time shaded circle or shaded cross
symbol used or masking. The symbols are joined up by a solid line of the appropriate colour.
 For bone conduction (BC), the right ear is represented by < (unmasked) or [ (masked),
whereas the left ear symbol is > (masked) or ] (masked). BC results are joined by a dotted
line.
 When there is no response at the highest output for the frequency under test, an arrow
pointing downwards from the appropriate symbol is drawn to indicate no response. The
symbol is drawn at the maximum level tested.
Right Ear Left Ear
 Uncomfortable loudness level is shown by an ‘L’ drawn to face towards the test side ( └ for
right ear and ┘for left ear).
 Another symbol used when testing is performed through the speakers or in the sound field
is S. This would represent the response of at least one ear or the response of the better
hearing ear. Symbol for aided audiogram, representing the hearing level with amplification
with hearing aids (A) or cochlear implants (C or CI).
Audiogram interpretation
 The hearing threshold is the sound level below which a person's ear is unable to detect any
sound.
 Hearing thresholds are computed using a decibel scale relative to the hearing thresholds of
normal hearing listeners, referred to as decibel hearing level (dB HL).
 The audiogram results must be looked at for each frequency in terms of

(1) the amount of hearing loss by air conduction (the hearing level),
(2) the amount of hearing loss by bone conduction, and
(3) the relationship between air-conduction and bone-conduction thresholds.
 A basic audiogram will show thresholds by air and bone conduction. The results should be
interpreted in terms of the:
 Degree of Hearing Loss: Degree of hearing loss is used to classify the magnitude or amount
of hearing loss. The degree of hearing loss is often categorized according to the pure tone
average (PTA), which is the average of the hearing thresholds at 500, 1000, and 2000 Hz, the
most important frequencies for understanding speech.
Normal hearing –10 to 15 db HL

Slight hearing loss 16 to 25 db HL
Mild hearing loss 26 to 40 db HL
Moderate hearing loss 41 to 55 db HL
Moderately severe hearing loss 56 to 70 db HL
Severe hearing loss 71 to 90 db HL
Profound hearing loss 91+ db HL
 Shapes of hearing loss:

There are four shapes of hearing loss
1) Bilateral means hearing loss in both ears.
2) Unilateral means hearing loss in one ear.

3) Symmetrical means the degree and configuration of hearing loss are the same in each
ear.
4) Asymmetrical means degree and configuration of hearing loss are different in each ear.
Symmetric hearing loss is typically defined as hearing in which pure-tone air conduction
thresholds in the right and left ear are separated by ≤10dB HL across all frequencies.
One interpretation of asymmetric hearing loss that may warrant a referral to an otologist is
two pure-tone air conduction thresholds with a difference of ≥15dB HL or one threshold with
a difference of ≥20 dB HL.
 Types of Hearing loss: In general terms, there are two types of hearing loss, conductive and
sensorineural. A combination of both is also seen as a mixed hearing loss.
1. Conductive Hearing Loss: This type of hearing loss involves the outer and/or middle ear.
It can be caused by many factors, such as a cerumen impaction, fluid in the middle ear, or
disarticulation of the bones of the middle ear.
On the audiogram, this type of hearing loss is characterized by having an air—bone gap that
is >10 dB HL and bone conduction thresholds that are within the normal range of hearing.
Audiogram depicting a right ear with mild conductive hearing loss.
2. Sensorineural Hearing Loss: This is the most common type of hearing loss that is seen
clinically. A sensorineural hearing loss indicates hearing loss that involves the inner ear
and/or the eighth (auditory) nerve. It can be caused by many factors, such as normal aging
of the hearing system, genetics, or noise exposure.
On the audiogram, this type of hearing loss is characterized by having an air-bone gap that
is ≤ 10dB HL (the bone conduction and air conduction thresholds are within 10dB HL).
3. Mixed Hearing Loss: This type of hearing loss involves the outer and/or middle ear and
the inner ear and/or eighth nerve. It can be caused by a combination of factors already
mentioned.
On the audiogram, this type of hearing loss can be characterized by having both the
aforementioned characteristics and/or having an air—bone gap >10dB HL, but the bone
conduction thresholds are not within the normal range of hearing.
Audiogram depicting a right ear with sensorineural hearing loss.
Audiogram depicting a right ear with mixed hearing loss.

AUDIOMETRIC CALIBRATION
 To ensure that an audiometer is performing in accordance with the relevant standard, the
instrument’s electroacoustic characteristics are checked and adjusted as necessary, usually
following a routine procedure.
 These calibration activities may be conducted at the manufacturing facility or an outside

laboratory, but are most often accomplished on site at least annually.
 In addition, records of calibrations must be maintained and dates of calibrations should

become part of the permanent audiometric record.
Types of calibration
1. Biological (Human) Calibration
2. Electroacoustic Calibration
1. Biological (Human) Calibration
 The person with primary responsibility for the audiometric testing should perform biological
checks.
 These checks should be done each day prior to client testing or anytime during the day when
there is reason to suspect that the audiometer is not performing properly.
 The biological check consists of two elements:

1) The self-listening check: The listening check involves listening for static, hum, noise,
crackling and popping sounds, and sound intensity changes while gently wiggling all wires
attached to the audiometer.
Presence of any of these undesired sounds requires further investigation and
remediation. If remediation is not successful, a qualified service technician should be
consulted.
2) The hearing threshold check: The hearing threshold check should involve threshold
measurement of a person with known and stable hearing thresholds. As a minimum, a
hearing threshold on the better ear of the listener should be performed with right (red)
and left (blue) earphones.
If measurements do not differ by more than ± 5 dB from each other and from the known
threshold of this person, the audiometer may be considered in calibration. If they differ
more than ± 5 dB, the audiometer needs an electroacoustic calibration.
2. Electroacoustic Calibration
 Technical, instrument-based assessment of an audiometer is typically referred to as

electroacoustic calibration.
 Electroacoustic calibration should be undertaken at least once a year or more frequently if

circumstances dictate.
1) Sound Pressure Measurement

Earphones
 Calibration of an audiometer is fundamentally the determination of the sound pressure

produced by the earphone of the audiometer.
 Sound pressure levels are best checked by an 'artificial ear' or coupler together with a sound
level meter. The artificial ear consists of a condenser microphone and coupler.
 A device that is used for the calibration of earphone output is the acoustic coupler. The
coupler consists of a heavy brass shell, enclosing a specified volume of air in an enclosure of
simple geometric design.
 The volume of air within the coupler is about 5.7 cc, or approximately the volume within an
average ear canal when an earphone is placed over the ear.
 The earphone is placed over an opening in the coupler, and the sound pressures which it
produces in the air volume are measured by means of a calibrated pressure-sensitive
microphone.
 Couplers serve as the connecting interface between the earphone and the measuring
instrument, which in this case is the sound level meter.
 The sound level meter has a microphone that detects minute fluctuations in air pressure.
Sound level meter measures the output SPL in decibels.
Loudspeakers
 As with earphones, the test signals presented into the test room via a loudspeaker need to
be calibrated. No coupler is involved in the calibration of loudspeakers and no specific type
of loudspeaker is recommended in the standards.
 A sound level meter is mounted vertically on a sturdy tripod and oriented so that the
microphone is at the reference point. It is important to note that a free-field-type
microphone pointed toward the sound source at a 0-degree angle of incidence is used for
sound-field calibration.
 The reference point must be at-least 1m away from the front of the loudspeaker and the
subject is not present in the test room during calibration. More specifically, the reference
point is located approximately where the center of the subject’s head would be during
testing.
2) Attenuator linearity
 Attenuator linearity refers to the change in output level of the audiometer as the HL dial is
manipulated.
 A multimeter is used to measure attenuator linearity. The output of the audiometer is

connected to the input of the multimeter, which is the preferred instrument to use. This
electrical connection is made with a Y-cord, with the multimeter connected to one arm of
the branch and the earphone attached to the other arm.
 Because attenuator performance is the focus, the multimeter is inserted between the output
of the audiometer and the earphone. The multimeter may display the measured output in
decibel units.
3) Frequency Accuracy
 After checking the performance of the audiometer’s attenuator, the accuracy of its pure
tone generator must be assessed.
 A frequency counter is used to assess the accuracy of an audiometer’s pure tone generator.
 The setup is similar as the one used to evaluate attenuator linearity. The same Y-cord is
connected to output of the audiometer while one branch of the Y is connected to an
earphone. The difference is a frequency counter replaces the multimeter in the other
branch.
 The attenuator control is set to 70-dB HL to ensure a strong signal. The frequency associated
with each of the frequency control settings is measured. The measured frequency is read
from the digital display in a straightforward way.
4) Harmonic Distortion
 Harmonic distortion refers to the presence of frequency components at harmonics (i.e.,

integer multiples) of the test frequency. For example, if the test frequency is 500 Hz, the
presence of frequency components (harmonics) at 1000, 1500, 2000 Hz, and so on is
indicative of harmonic distortion.
 Because the audiometer and/or the earphone can generate distortion, it is necessary to
make measurements at the output of the earphone.
 The earphone is placed on the coupler, and its acoustic output is picked up by the condenser
microphone, which is connected to an amplifier and a wave analyzer. The amplitudes of the
various harmonics are measured by means of the spectrum analyzer.
 An added benefit of the spectrum analyzer is that because it displays a range of frequencies,
it is possible to determine whether the audiometer is generating significant harmonic
distortion.
5) Signal Switching
 The pure tone signal is delivered to the listener by pressing the interrupter switch. Three
factors associated with the function of the switch are checked during calibration: on–off
ratio, crosstalk, and rise and fall time.
 The on–off ratio describes the output level of the signal when the tone is on (i.e., the switch
is pressed or closed) versus when the signal is off (i.e., the switch is released or open).
 The oscilloscope has a screen for display purposes. The scope shows the waveform, which is
the output level as a function of time.
 The scope is especially helpful for examining a pure tone signal as it is turned on and off. The
time it takes for the signal to go from completely off to on is known as the rise time. The fall
time is just the opposite.
 The rise and fall times cannot be too abrupt or audible transients may occur. These audible
transients contain frequencies other than the test frequency and their presence reduces the
specificity of the hearing test.
 Because the audiometer’s switch reacts very quickly, an analog storage oscilloscope or a
digital oscilloscope is required to capture the on-off events accurately.
AUDIOMETER SYSTEM BEKESY
 George Van Bekesy, a Hungarian scientist, designed an automatic audiometric testing

method for plotting the hearing threshold based on the patient’s signal.
 A principal feature of the method, differentiating it from conventional pure-tone
audiometric techniques, is the interdependence of the patient’s response and stimulus
intensity: responses govern intensity and are affected by changes they introduce in it.
 An audiogram traced by the Bekesy method represents the absolute threshold values at all
frequencies in the range tested. In addition, it shows the difference, in decibels, between
levels at which the patient just hears a signal of increasing intensity and those at which he
just ceases to hear the signal when its intensity is decreasing.
 This latter characteristic often varies significantly with the type of hearing impairment, and
can aid in establishing the site of lesion within the auditory system.
 On the basis of the audiograms, one can easily separate the conduction and perceptive
hearing deficiencies from each other.
 Audiometers Bekesy are relatively simple for the patient to operate. The instrument
generates a pure-tone signal, which is presented to him through an air-conduction
earphone. The subject is told to press a switch when the tone is heard and to release the
switch when it is not heard. This switch controls the motor-driven attenuator of the
audiometer: when it is pressed, signal intensity decreases and when it is released, signal
intensity increases. A pen connected to the attenuator traces a continuous record of the
patient’s intensity adjustments on an audiogram chart, producing a graphic representation
of the subject’s threshold. The test signal may be presented in a variety of ways, each suited
to the investigation of a particular problem.
 A block diagram of the audiometer system Bekesy is shown in Fig. It consists basically of an
electrical section and a mechanical section. The electrical section includes an oscillator and
modulator circuits for the generation of the desired test signal, an automatic attenuator
linked to the writing system, control circuits for the drive motors of the mechanical section
and a master clock generator for the control of all timing functions via a logic control circuit.
The carriage drive and the writing system with their separate drive motors constitute the
mechanical section.
Block diagram of the audiometer system Bekesy
Electrical Section
 Sine Wave Oscillator: This oscillator generates test signals with frequencies of 125, 250, 500,
1000, 1500, 2000, 3000, 4000, 6000 and 8000 Hz. This sequence is first presented to the left
ear automatically, each tone for 30 s, and then to the right ear, the shift between the
frequencies being noiseless. After both ears have been tested, a 1 kHz tone is presented to
the right ear to provide a useful indication of test reliability.
 Modulator: From the oscillator the test signal is fed to the modulator, where the mode of
operation is selected by the ‘Tone’ switch, via the logic control circuit. Two models, ‘Pulse’
or ‘Cont’, are available. In the ‘Pulse’ mode the test signal is modulated giving a signal, which
is easily recognized by the patient. In the ‘Cont’ mode no modulation is applied, giving a
signal suitable for use, when calibrating the audiometer.
 Automatic Attenuator: The signal from the modulator feeds the automatic attenuator
situated on the carriage together with the writing system. The attenuator consists of a
logarithmic potentiometer which has its wiper attached to the pen drive so that the
attenuation of the potentiometer corresponds to the position (y-axis) of the pen on the
audiogram chart. The potentiometer has infinite resolution. The attenuation range is 100 dB,
thereby covering the range of hearing levels from -10 to +90 dB. When the test is initiated,
the attenuator starts at its top position of -10 dB and then increases the level with a rate of
5 dB/s. Also, when the test signal switches between the ears and when retesting at 1 kHz,
the attenuator decreases the signal level to –10 dB to ensure that the right ear does not
receive a tone at the elevated level possibly required at 8 kHz.
 Hand Switch: The pen drive is controlled via the logic control circuit by means of the hand-
switch operated by the patient. Pressing the switch decreases the output from the
potentiometer and thereby the level in the earphones, while releasing the switch increases
the output both ways with a speed of 5 dB/s.
 Buffer Amplifier and Calibration Circuit: From the attenuator the signal is fed via a buffer
amplifier to the hearing level calibration circuit. The buffer amplifier isolates the attenuator
from the calibration circuit in order not to affect its output. The calibration circuit consists
of seven potentiometers, one for each test frequency. During calibration, the
potentiometers are adjusted one at a time until the correct level, measured in a coupler, is
obtained in the earphones.
 Earphones: The earphones are a matched pair with distortion, typically less than 1%.
 Master Clock Generator: A stable clock generator supplies the necessary signals for the
control of motor speed, attenuator speed, frequency shift, modulation and other timing
functions. This makes the system independent of variations in line voltage and frequency.
Mechanical Section
 Carriage: The carriage with the writing system is driven by a stepping motor via a toothed
belt. The speed and direction of rotation of the motor are automatically controlled via the
logic control system. When the test is initiated and the patient indicates that he hears the
signal by pressing the hand switch, the carriage moves along the X-axis (frequency axis) of
the audiogram in agreement with the frequency of the test signal. When the frequency
shifts, the carriage stops until the patient again, by pressing the hand switch, indicates that
he hears the signal. This avoids wastage of recording space on the audiogram if a patient’s
hearing threshold varies from frequency to frequency or from left to right ear. When the
complete test is finished the carriage and writing system return to the start position. To
prevent carriage over-run, two limit switches are included in the carriage drive circuit.
 Writing System: The writing system is operated by the pen drive, which is driven by a
stepping motor. The pen drive moves the pen, and with it the wiper of the automatic
attenuator, along the Y-axis (hearing level axis) with a constant speed corresponding to the
change in attenuation of 5 dB/s. The direction of movement of the pen is determined by the
position of the hand switch operated by the patient. Limit switches are also included with
the pen drive.
 Audiogram Chart: The audiogram is printed in standard A5 format (148 ¥ 210 mm). The
recording space is large, 0.8 dB/mm, to enable easy reading. Space is provided on the
audiogram side for registration of information on the patient, audiometer, operator, etc.
while the other side has space for recording the patient’s medical and occupational history.
Four holes in the chart give precise and automatic location of the audiogram on the chart
bed.
 In order to establish a more exact diagnosis applying adaptation and hearing fatigue tests,
several other tests besides the pure-tone Bekesy audiometry, have been suggested and can
be performed using the basic Bekesy system. For example, for carrying out the Fowler
loudness balance test, a second channel is provided. The second channel has a continuously
variable intensity over the range 0 to 110 dB and is calibrated in 1 dB increments.
EVOKED RESPONSE AUDIOMETRY SYSTEM

 For every sound presented to the ears there will be a corresponding electrical evoked
response resulting from the activity of the cochlea, the cochlear nerve, and the brain. These
evoked responses can be recorded and used to investigate a person’s hearing.
 Evoked response audiometry has been the subject of research for several years. This work
has established evoked response electroencephalography resulting from an auditory
stimulus above the hearing threshold.
 Instruments based on this principle have been found particularly suitable for determining
auditory threshold in the absence of voluntary response in subjects such as infants,
uncooperative adults, or animals.
 The system basically comprises a conventional wide range pure-tone audiometer, which
operates under the control of an automatic programmer and provides a series of auditory
stimuli to the subject via either a loudspeaker or standard earphones.
 The EEG signal is picked up by standard electrodes placed in contact with the subject’s scalp.
One electrode is usually placed on the vertex, one at the post auricular area, and a third
(ground) on the earlobe or forehead.
 The instrument stores and evaluates that part of the EEG signal, which follows each
individual stimulus presentation. At the end of the programmed series of stimuli, it writes
out on a paper chart a waveform that is the average response to stimuli.
 The presence of characteristic amplitudes and latencies in this waveform give an indication
that the test intensity exceeded the subject's threshold at the test frequency. Similar trials
at other intensity levels and other frequencies establish the threshold contour.
Block diagram of the evoked response audiometer
 Evoked response audiometry system consists of the following five major subsystems.
The Tone Generator:
 It is a wide range pure-tone audiometer whose frequency output can be selected at 250,
500, 750, 1000, 1500, 2000, 3000, 4000, 6000 and 8000 Hz. The output power levels are
adjustable from -5 to + 110 dB in 5 dB steps.
 In addition to the pure tones, internally generated broad-band noise may be used as the
stimulus. Provision is also made for external input from other types of stimulus. A special
feature of the generator is the selectable rise/decay times for 1-100 ms.
 Outputs are provided for the left ear, right ear, or both. A variable intensity masking noise
source is included in the generator. A power amplifier is incorporated to drive a speaker or
tactile transducer. Total harmonic distortion should not exceed 2%.
EEG Amplifier:
 It is a conventional high gain, high impedance, low noise amplifier. The first stage of the EEG
amplifier is preferably kept in a separate "preamp head" located near the subject. Its design
and location minimize power line frequency pick-up.
 An ohm meter provided in the Preamp head enables the measurement of the electrode
contact impedance and thus indicates when satisfactory contact resistance of the electrodes
is obtained. The EEG preamplifier gain is fixed at 200. The overall sensitivity is adjustable in
steps of 10 to 1000 µV/div on the chart recorder.
 The amplifier also provides selectable roll-offs at the high and low ends of the spectrum of
interest and a 60Hz sharp notch filter. The low frequency roll-off points are at 0.15, 0.30.0.60,
1.5, 3, 6, 10, 15, 30, 60, at 6dB (half amplitude), whereas high frequency points are 1.5, 3, 6,
10, 15 ,30, 60, 100 Hz at 6 dB (half amplitude) yielding a 12 dB / octave roll-off.
The Programmer:
 A logic device that controls the system operation in correct time sequence. It helps to have
a selectable rate of stimulus presentation, stores the number of pulses that the operator
chooses to constitute a run, starts the recorder at the beginning of the run, turns the
audiometer tone generator on and off to provide the auditory stimuli at the proper time.
 It also speeds up the chart drive for the detailed signal samples, stops the recorder after
providing for paper clearance, erases the signal averaging computer and clears and resets
itself for the next run.
 Total count in the programmer is selectable from 1-109 stimuli with a selector switch. The
pulse interval is normally kept as 0.1, 0.2, 0.5, 1, 2, 5, 10 and pulse duration as 1-10,000 ms.
Signal Averaging Computer:
 This separates evoked responses from the normal EEG activity by ignoring those
components, which are not synchronized with stimuli. Because the waveform of the evoked
potential will be essentially the same every time in response to the tone presentation, and
the other electrical activity will vary randomly, the evoked response " grows" in the
computer memory and the noise component tends to average to zero with repeated
presentations. It may be provided with either 50 or 100 averaging points depending upon
the degree of resolution required.
 The computer includes a provision for selection of integrating time constant (5, 10, 20, 50,
100, 200, 500 s) and sweep duration or analysis time (0.1, 0.2, 0.5, 1, 2, and 5 s). A delay
circuit is incorporated to select a delay between the onset of the stimulus and the start of
analysis by the signal averager.
 The delay time is selectable as 0, 0.2, 0.5 and 1 s. The amplitude of the evoked response may
be normalized to that of the on-going EEG monitoring signal using a gain control (gain
variable from 20 to 50). The computer provides outputs for display either on a conventional
oscilloscope or on an X-Y plotter.
Chart Recorder:
 It is a two channel recorder. One of the two channels is used to display the averaged
response after it has been processed by the computer and the other displays unprocessed
EEG. There are two event markers, one of which is activated by each gating pulse from the
programmer to show the beginning and duration of each stimulus and the other is available
for registering any mark at the desired instant.
 The chart can be driven at four different speeds (1, 5, 25, 125 mm/s), which are automatically
switched by the programmer. Translucent chart paper is usually employed so that records
may be compared by overlaying one on another on the illuminated opal-glass viewer.
 Evoked response audiometer systems also contain a provision for 'External' mode of
operation where any other type of stimulus generator can be connected into the system and
controlled from the programmer. This could include narrow band noise, speech, a high
frequency auditory signal for animal research or even a photic or tactile stimulator.
 Modern evoked response audiometers are built around microcomputers. In these
instruments, the stimulators, preamplifiers and amplifiers are all digitally controlled via the
central processing unit, which automatically avoids undesirable parameters. They have no
push button or dials as the parameters are varied by means of the keyboard.
 The parameters can be controlled to a very wide range, which would not be possible with
conventional knobs and switches. Stimuli from 12 to 16kHz are included to facilitate
investigation into high frequency hearing loss and ototoxic drug effects. Texts and
waveforms are displayed on a large size TV screen. A built-in chart recorder helps to make
recordings under the control of the keyboard.
HEARING AIDS
 A hearing aid is an electronic device for processing and amplifying sounds to compensate for
hearing loss. The primary objective in hearing aid amplification is to make all speech sounds
audible, without introducing any distortion or making sounds uncomfortably loud.
 The basic components of a contemporary hearing aid include a microphone, an amplifier, a
receiver, and a power supply.
TYPES OF HEARING AIDS

 Hearing aids are signal processors: That is, they alter the signal input to improve it for the
wearer.
Conventional Hearing Aid

 Analog hearing aids, so called because the electrical signals generated are analogous to the
sound that comes into the instrument.
 Modern hearing aids have evolved from single-transistor amplifiers to modern multi-channel
designs containing hundreds and even thousands of transistors.
 The basic functional parts include a microphone and associated preamplifier, an automatic
gain control circuit (AGC), a set of active filters, a mixer and power amplifier, an output
transducer or receiver. The total circuitry works on a battery.
 The use of multiple channels in this design provides different compression characteristics for
different frequency ranges. Typically, the crossover frequencies of the channels and the
compression characteristics can be adjusted with potentiometers. Most of the latest hearing
aids are electronically programmable.
 The programmable parameters are downloaded from a computer-based system and stored
in digital registers. The register outputs are used to switch resistor networks that control
various analog circuitry.
 The active filters are adjusted to generally provide for low-frequency attenuation of up to
30-40 dB relative to the high-frequency response. This is because most hearing aid wearers
require high frequency gain.
 The transducer in a hearing aid, which is a microphone, can be realized in an integrated form
with a field-effect transistor preamplifier. The preamplifier is housed in the metallic,
microphone case to shield its input from extraneous noise. On the other hand, the receiver
is an electromagnetic device, which drives a miniature diaphragm to produce acoustic
output.
 The acoustic output is routed to the ear-mould through a flexible tubing whose frequency
response can be altered to boost the high-frequency response. This is done by tapering its
inside diameter from the ear mould back to the receiver port end.
Conventional Analog Type Hearing Aid
 All the electronics circuitry is packaged in a housing, which can be designed for fitting to the
ear in any one of the following ways:
1. Placing all the components in a pocket-sized enclosure or box which is connected to the
output transducer worn in the ear. The box can be carried in the shirt pocket or carried
with a belt around the waist. With the availability of miniature-sized aids, this approach
is no longer employed.
2. The components are packaged in a curved module, which is designed to fit comfortably
behind the ear.
3. The most popular design is in which the total package can be put inside the outer ear.
Integrated Microphone and PET Amplifier
 Much depends on the performance of the filters for further reduction of the size and
improvement in the working of the hearing aids. The dynamic range of an operational
amplifier, which is the basic building block of an electronic filter, decreases as the three
halves power of feature size. Since dynamic range of analogue hearing aids is already
marginally acceptable, it appears that further reduction in size to achieve increased
processing complexity is not practical. The potential for greater dynamic range, with less
power consumption and greater complexity in hearing aid design is feasible only with digital
processing technologies.
Digital Hearing Aid
 Digital programmable hearing aids have all the features of analog programmable aids but
use digitized sound processing (DSP) to convert sound waves into digital signals.
 The major parts are the microphone, an analog-to-digital converter (ADC), the digital signal
processor (DSP), the digital-to-analog converter (DAC), the receiver and a two port memory.
Essentially, sound waves picked up by the microphone and transformed into electrical
signals are converted into digital form by an A-D converter.
Block Diagram of Digital Hearing Aid
DAC = Digital-to-analog converter REC = Receiver MIC =

Microphone
ADC = Analog-to-digital converter DSP = Digital signal processor
 A typical microphone will have an internal noise of 20 dB SPL (sound pressure level) when
referred to the input and maximum undistorted output corresponding to a signal of about
90 dB SPL. Allowing some margin for peak performance, the total dynamic range required of
the ADC is 80 dB. This requirement can be achieved with a 14bit A-D converter.
 The DSP is a fixed (wired-program) digital processing device containing an array of adders,
multipliers and registers which provide the fundamental operations necessary for
implementing various digital algorithms. hr a general-purpose DSP, considerable power is
consumed in executing the programme instructions. Since power consumption is a major
consideration in the design of hearing aids, the wired-program approach is followed. The
DSP is associated with a two-port memory, which is used to store processing parameters
that can be down loaded from the external programmer to the hearing aid while it is
adjusted for the intended user.
 The dynamic range requirements of the DAC are more severe. Some hearing impaired
listeners have almost normal sensitivity at low frequencies but significantly elevated
thresholds at high frequencies. Since the conversion noise generated by the DAC has a
uniform spectrum and is a function of the overall output signal level, high-level high-
frequency sounds can create low- frequency noise and distortion that falls above the
threshold at low frequencies.
 The digital hearing aids are implemented with CMOS technology, with a feature size of 1 µm
or less and with an estimated power consumption of 20 µW. An estimated 10,000 CMOS
inverters are required to implement 400,000 multiply-add operations for filtering,
compression functions and other processing requirements.
 The digital hearing aids promise to provide capabilities of superior signal processing, ease of
fitting and stable long-term performance. However, they are still under development. It has
often been seen that a person buys a hearing aid but does not use it because it does not help
very much. The basic reason is that the impaired ear has its capacity to process speech and
hearing aids are simply sound amplifiers that do not compensate for the loss of processing
power. It needs to be emphasized that today's hearing aids are at an early stage of
development and need to reach a highly refined stage before they can find wide spread and
useful applications. The potential areas of improvement include shaping the frequency
response to invert the patient's hearing loss, enhancing the signal-to-noise ratio with
adaptive filtering, reducing acoustic feedback and compressing/expanding signals with

minimum distortion.
 This digital technology is the most expensive, but it offers many advantages. Key benefits
include:
 improvement in programmability
 greater precision in fitting
 management of loudness discomfort
 control of acoustic feedback (whistling sounds)
 noise reduction
COMMON HEARING AIDS

 Conventional hearing aids can be divided into two types:
1. Air conduction hearing aid: In this, the amplified sound is transmitted via the ear canal to
the tympanic membrane.
2. Bone conduction hearing aid: Instead of a receiver, it has a bone vibrator which snugly fits
on the mastoid and directly stimulates the cochlea.
Some people are unable to use air conduction hearing aids due to chronic ear infections or
malformed ear canals. For these individuals, a bone conduction hearing aid is most
appropriate.
The most common type of bone conduction hearing aid is a BAHA (Bone Anchored Hearing
Aid). Inputs to this type of hearing aid are converted to mechanical vibrations that shake
the skull and stimulate the receptors in the cochlea, BAHA take advantage of the fact that,
at a sensory level, it does not matter whether sounds come from an air conducted hearing
aid or a bone conducted hearing aid.
 Most the aids are air conduction type. The range of hearing aid types is very wide but they
can be classified according to where they are worn. Five major categories are:
1. Body worn.
2. Behind the ear (BTE).
3. In the ear (ITE).
4. In the canal (ITC).
5. Completely in the canal (CIC)
Body-worn aids
 Body-type hearing aids are rarely used today, being reserved for patients with very profound
hearing losses. The first wearable electronic hearing aid was the body hearing aid.
 These instruments contain the microphone, amplifier, circuit modifiers, and battery
compartment within a case that may be clipped to the wearer’s clothing or worn in a pocket
or a special pouch. A cord carries the electrical signals to a receiver, which is coupled to the
patient’s ear through a custom-fitted earmold.
 Body-worn aids can be relatively larger which enables high-quality components and large
batteries to be used. For these reasons the body-worn aid usually gives the largest
‘maximum output’ and the best sound quality.
 However, behind-the-ear and within-the-ear aids are usually more acceptable than the
body-worn aids because they are more convenient and more socially acceptable.
Behind-the-Ear (BTE) Hearing Aids

 Behind-the-ear (BTE) or postaural hearing aids are those in which all the components are
housed in a case that is situated behind the pinna and linked to the meatus by a tube and
earmould.
 It is useful for slight to moderate cases of hearing loss particularly the high frequency ones.
 The relatively large size allows complex circuitry to be included, but they are more visible
than an ITE aid.
 The fact that the microphone is outside the pinna means that no natural ear resonance
effects are obtained.
Open-Fit Hearing Aids

 These aids are often significantly smaller than traditional BTE instruments and are coupled
to the ear with a very thin tubing, which adds to their cosmetic advantage.
 The open-designed ear tip, which frequently incorporates the hearing aid receiver within the
ear canal, provides a more comfortable fit largely free of what has become known as the
occlusion effect, the booming sound many hearing aid wearers complain about when
listening to their own voices.
 These open-fitting hearing aids provide excellent sound quality, better directional
microphone placement, and less occlusion compared with many ITE hearing aids.
 The earpiece of an open fit hearing aid is a small, soft rubber or silicone cap, known as a
dome. This type of aid is normally suitable for patients with mild or moderate hearing loss.
 The fitting time is usually shorter because earmold impressions are not needed, because the
dome fits into the ear canal.
In-the-ear hearing aids

 In-the-ear (ITE), also occasionally called intra-aural, hearing aids are those in which all the
components are housed within the ear fitting itself, which is known as the shell of the aid.
‘In-the-ear’ is a generic term used to refer to the whole range of aids worn within the pinna.
Quite high gain is possible without feedback if the aid fits well into the ear. The aids are
quite visible, however, and the microphone can be subject to wind noise. The maximum
output is lower than that of a BTE aid.
 Originally useful only for mild hearing losses, because of improved technology they can be
used for hearing losses that range from mild to moderately severe.
In-the-Canal (ITC) Hearing Aids

 The ITC aids primarily fits within the concha and in the outer half of the canal and extend
down into the ear canal. They have the advantage of being less visible than other aids, a
reduced effect of wind noise and the ability to use natural resonances within the ear.
 The faceplate of this aid is accessible to the user to allow changing the volume control and
turning the aid on and off. This aid provides some advantage in gain at higher frequencies
because of its depth of insertion and the acoustic resonance in the unblocked concha.
Disadvantages include short battery life and an increased chance of feedback at higher gains.
Completely-in-the-canal (CIC) hearing aids

 Completely-in-the-canal (CIC) hearing aids fit fully within the ear canal. It is the smallest of
all hearing aids. It typically fits entirely within the ear canal, and the deepest portion of the
aid is in close proximity to the tympanic membrane.
 The faceplate is usually not accessible to the user. The aid also needs a short cord or wire
attached to the faceplate for the wearer to use while removing the aid.
 These aids are regarded as the most cosmetically pleasing, and, because of the close
proximity to the tympanic membrane, they can reduce or eliminate the occlusion effect or
the barrel sound experienced by many hearing aid users. Additionally, individuals with this
type of aid can use the telephone like those without hearing aids. Because of the size
limitations of the internal components of the aid and the physical limitations of the ear canal
itself, these hearing aids are most appropriate for individuals with mild to moderate hearing
loss.
COCHLEAR IMPLANT
 A cochlear implant (CI) is a surgically implanted electronic device that provides a sense of
sound to a person who is profoundly deaf or severely hard of hearing. The cochlear implant
is often referred to as a bionic ear.
Introduction
 Sensori-neural deafness affects a large number of people throughout the world. The
treatment of choice for the sensori-neural deaf is the cochlear prosthesis or cochlear
implants.
 Sensori-neural deafness can be caused either by cochlear damage or by damage within the
auditory nerve or to the neurons of the central auditory system. The hair cells are the sensory
cells that transduce mechanical motion into signals that can be recognized by auditory
neurons. The auditory neurons carry information from the hair cells to the cochlear nucleus
in the brainstem and, via the cochlear nucleus, to higher nuclei in the brain.
Details of Cochlear part of Human Ear
 The normal cochlea and the associated neurons of the central auditory system provide
information about both the frequency content and intensity of the auditory signal.
Information is conveyed to the acoustic nerve about frequency content by the mechanically
tuned properties of the basilar membrane.
 The inner hair cells, which connect to the vast majority of afferent neurons, are thought to
be the sensory cells of the cochlea whereas the role of the outer hair cells is still under
investigation.
 The location of hair cells along the cochlea determines their optimal response to frequency:
hair cells at the apex are responsive to low frequencies, while hair cells at the base are
responsive to high frequencies. The distribution of frequencies along the spiral is logarithmic.
Diagram of the basilar membrane showing the base and the apex. The position of maximum
displacement in response to sinusoid of different frequency (in Hz) is indicated
 If the damage to the auditory system is peripheral in the inner ear, then a cochlear implant
can be used' In the general design of a cochlear implant, the sound is decomposed into
frequency bands of use for the transmission and reception of speech, and critical features of
the signals within those frequency bands are delivered to auditory neurons via an array of
electrodes.
Parts of the cochlear implant
 The implant is surgically placed under the skin behind the ear. The basic parts of the device
include:
External parts:
 One or more microphones which picks up sound from the environment
 Speech processor which selectively filters sound to prioritize audible speech splits the sound
into channels and sends the electrical sound signals through a thin cable to the transmitter,
 A transmitter, which is a coil held in position by a magnet placed behind the external ear,
and transmits power and the processed sound signals to the internal device by
electromagnetic induction,
Internal parts:
 A receiver and stimulator secured in bone beneath the skin, which converts the signals into
electric impulses and sends them through an internal cable to electrodes,
 An electrode array is a configuration of electrodes. An array of up to 24 electrodes wound
through the cochlea, which send the impulses to the nerves in the scala tympani and then
directly to the brain through the auditory nerve system.
There are 4 manufacturers for Cochlear implants, and each one produces a different implant
with a different number of electrodes. Advanced Bionics produces implants with 16
electrodes and uses a technique called current steering in which two electrodes are
stimulated simultaneously with different current levels to produce intermediate virtual
channels. The number of channels is not a primary factor upon which a manufacturer is
chosen; the signal processing algorithm is also another important block.
Instrumentation
Block diagram of a generic cochlear implant
 The microphone converts acoustic signals into electrical signals. The electrical signals are
amplified and encoded in various ways in the block called stimulus encoder. In the vast
majority of implants, the stimulus encoder is worn outside the head, producing a serially
coded signal that is transmitted with a transcutaneous link, most often inductive.
 The link sends both data and power to an internal circuit that decodes the serial data stream
and decomposes it into signals that are delivered to the current sources that drive the
electrodes of the cochlear electrode array. Each electrode of the array is driven with either
a pulsatile or an analogue electrical signal. The signals traverse the tissues of the inner ear,
usually the fluids of the scala tympani, and excite the auditory neurons. The excitation
depends upon the number of intact neuron that remain, the proximity of the electrode array
to the neurons/ and the spatial and temporal characteristics of the current-density fields
that affect the neurons.
 In single channel implants, only one electrode is used. In multi-channel cochlear implants,
an electrode array is inserted in the cochlear so that different auditory nerve fibres can be
stimulated at different places, thereby exploiting the place mechanism for coding
frequencies. Different electrodes are stimulated, depending or, the frequency of the signal.
 Electrodes near the base of the cochlea are stimulated with high frequency signals while
electrodes near the apex are stimulated with low frequency signals. The signal processor is
responsible for breaking the input signal into different frequency bands or channels and
delivering the filtered signals to the appropriate electrodes.
 Different types of cochlear implants are available which differ in the following
characteristics:
o Electrode design: Number of electrodes and their configurations
o Type of stimulation: Analog or pulsatile
o Transmission link Transcutaneous or percutaneous
o Signal processing: waveform representation or feature extraction
 Cochlear implants have been a spectacular success story for biomedical engineers. They have
been successful in restoring partial hearing to profoundly deaf people. While not all patients
are able to talk on the telephone when they use their implants a substantial number can,
and all users can improve their communication skills by using lip reading. Users can hear
environmental sounds such as automobile horns, knocks at the door and sirens. Despite the
maturity of today’s cochlear implants, there are exciting opportunities for bioengineers to
advance designs to provide better devices to the patients.

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AD: Audiometers & Hearing Aids - 1

AUDIOMETRS AND HEARING AIDS

 The following events are involved in hearing,

 The primary components are

 Non-test ear (NTE): Opposite or contralateral ear

Interaural attenuation with air conduction. Interaural attenuation with bone

 Several different kinds of masking noises are available on commercial pure-tone

 There are three types of masking noise:

threshold of masking is denoted M, in the British Society of Audiology’s

 Speech Testing: Speech discrimination or word recognition ability is scored as a percentage.

 It is important to use standard symbols on an audiogram to avoid confusion.

Right Ear Left Ear

 The audiogram results must be looked at for each frequency in terms of

Normal hearing –10 to 15 db HL

 Shapes of hearing loss:

2) Unilateral means hearing loss in one ear.

Audiogram depicting a right ear with mixed hearing loss.

 These calibration activities may be conducted at the manufacturing facility or an outside

 In addition, records of calibrations must be maintained and dates of calibrations should

 The biological check consists of two elements:

 Technical, instrument-based assessment of an audiometer is typically referred to as

 Electroacoustic calibration should be undertaken at least once a year or more frequently if

1) Sound Pressure Measurement

 Calibration of an audiometer is fundamentally the determination of the sound pressure

 A multimeter is used to measure attenuator linearity. The output of the audiometer is

 Harmonic distortion refers to the presence of frequency components at harmonics (i.e.,

AUDIOMETER SYSTEM BEKESY

 George Van Bekesy, a Hungarian scientist, designed an automatic audiometric testing

Block diagram of the audiometer system Bekesy

EVOKED RESPONSE AUDIOMETRY SYSTEM

TYPES OF HEARING AIDS

Conventional Hearing Aid

DAC = Digital-to-analog converter REC = Receiver MIC =

adaptive filtering, reducing acoustic feedback and compressing/expanding signals with

COMMON HEARING AIDS

Behind-the-Ear (BTE) Hearing Aids

Open-Fit Hearing Aids

In-the-ear hearing aids

In-the-Canal (ITC) Hearing Aids

Completely-in-the-canal (CIC) hearing aids

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