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Claims  |
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We claim:
1. A hearing aid comprising at least one input microphone, an output
receiver, and a signal transmission channel interposed between said
microphone and said receiver in which the improvement comprises a
programmable delay line filter and programmable signal limiter means
interposed between the input and output of said transmission channel in a
feedback path for said transmission channel, said filter being programmed
to impart to the hearing aid at least one response characteristic
effective to compensate for impaired hearing of the wearer of the aid.
2. A hearing aid as in claim 1 in which said filter comprises a delay line
having a multiplicity of taps, first memory means for storing
filter-response related coefficients therein, and means jointly responsive
to said delay line taps and to coefficients stored in said memory means
for providing an output having at least one response characteristic
effective to compensate for impaired hearing of the wearer of the aid.
3. A hearing aid as in claim 2 in which said filter comprises a digital
delay line and means for converting analog signals in said channel to
digital signals for application to said delay line.
4. A hearing aid as in claim 2 in which said filter comprises an analog
delay line having a multiplicity of taps, and means for periodically
sampling analog signals in said channel and supplying them to said delay
line.
5. A hearing aid comprising at least one input microphone, an output
receiver, and a transmission channel interposed between said microphone
and receiver, means controllable to impart different response
characteristics to said hearing aid, and controlling means responsive to
the level of speech signals in said transmission channel in excess of the
level of noise signals in said channel for automatically controlling said
controllable means to impart a selected one of said different response
characteristics to said hearing aid.
6. A hearing aid as described in claim 5 in which said controlling means
comprises speech detector means for determining when speech signals are in
said transmission channel, a plurality of bandpass filter means for
determining the noise frequency spectrum in said transmission channel, a
plurality of comparator means each response to said speech detector and to
respective bandpass filter means for indicating whether the speech level
in each said bandpass filter exceeds the noise level therein and for
actuating said controlling means to impart to the hearing aid a response
characteristic effective to compensate for impaired hearing of the wearer
of the aid at the noise levels obtaining in said channel.
7. A hearing aid as described in claim 6 in which said controlling means
also comprises means for rectifying a portion of the signals in said
transmission channel, a plurality of differently pre-biased comparators
connected to receive the output of said rectifying means, and means
responsive to the outputs of said comparators for generating a signal
representative of the average level of the signals in said transmission
channel for controlling said controllable means to impart to the hearing
aid a response characteristic effective to compensate for impaired hearing
of the wearer of the aid at the signal and noise levels obtaining in said
channel.
8. A hearing aid as described in claim 7 in which said controllable means
comprises a delay line having a multiplicity of taps, memory means for
storing predetermined filter response-related coefficients therein, and
means responsive to said controlling means for combining said delay line
taps and coefficients to constitute a delay line filter of predetermined
response characteristic in said transmission channel.
9. A hearing aid comprising at least one input microphone, an output
receiver, a signal transmission channel interposed between said microphone
and said receiver, and a programmable delay line filter interposed in a
forward path between the input and output of said transmission channel,
said programmable filter being programmed to effect substantial reduction
of acoustic feedback from said receiver to said microphone.
10. A hearing aid as described in claim 1 in which said programmable delay
line filter is programmed so as to effect substantial reduction of
acoustic feedback from said receiver to said microphone.
11. A hearing aid comprising at least one input microphone, an output
receiver, a signal transmission channel interposed between said microphone
and said receiver, and a programmable delay line filter interposed in a
forward path between the input and output of said transmission channel
said filter comprising a digital delay line having a multiplicity of taps,
means for converting analog signals in said channel to digital signals for
application to said delay line, and first memory means for storing
filter-response related coefficients therein, said digital delay line
comprising second memory means arranged to receive digital signals from
said analog-to-digital converting means, register means connected to
receive digital signals from said second memory means, first
digital-to-analog converter means connected to receive digital signals
from said register means, timing means controlling the transfer of digital
signals from said second memory means through said register means to said
first digital-to-analog converter means, second digital-to-analog
converter means connected to receive filter response related coefficients
from said memory means in timed relation to the transfer of digital
signals to said first digital-to-analog converter means and to supply
analog signals therefrom to said first digital-to-analog converter means
for combination therein with digital signals transferred thereto from said
second memory means, signal summing means connected to receive the output
of said first digital converter means, and sample and hold means connected
to receive the output of said signal summing means and for providing an
output having at least one response characteristic effective to compensate
for impaired hearing of the wearer of the aid.
12. A hearing aid comprising at least one input microphone, an output
receiver, a signal transmission channel interposed between said microphone
and said receiver, and a programmable delay line filter interposed in a
forward path between the input and output of aid transmission channel,
said filter comprising an analog delay line having a multiplicity of taps,
means for periodically sampling analog signals in said channel and
sampling analog signals in said channel and supplying them to said delay
line, and first memory means for storing filter-response related
coeffients therein, digital-to-analog converter means connected to receive
filter-response related coefficients from said first memory means for
conversion to analog values, multiplexer means connected to supply signal
samples from said sampling means to said digital converter means for
combination with said respective analog values, summing means for summing
said analog values and signal samples, and sample and hold means connected
to receive the output of said summing means and for providing an output
having at least one response characteristic effective to compensate for
impaired hearing of the wearer of the aid.
13. A method of reducing acoustic feedback in a sound system comprising a
microphone, a transducer and a signal transmission channel interposed
between said microphone and transducer, comprising the steps of
determining the effect on the amplitude and phase of a signal in said
transmission channel as a function of frequency of acoustic feedback
between said transducer and microphone, and
inserting between the input and output of said transmission channel an
electrical feedback path having a filter therein programmed to equalize
and reduce the effect of said acoustic feedback both in amplitude and
phase on a signal in said transmission channel.
14. A method of reducing acoustic feedback in a hearing aid comprising a
microphone, a receiver fitted in an ear of a wearer of the aid, and a
signal transmission channel interposed between said microphone and
transducer, comprising the steps of
determining the effect on the amplitude and phase of a signal in said
transmission channel as a function of frequency of acoustic feedback
between said receiver and microphone, and
inserting between the input and output of said transmission channel a
programmable filter programmed to equalize and reduce the effect of said
acoustic feedback both in amplitude and phase on a signal in said
transmission channel.
15. A method of reducing feedback in a hearing aid as described in claim 14
in which said programmable filter is inserted in a forward path through
said transmission channel.
16. A method of reducing feedback in a hearing aid as described in claim 14
in which said programmable filter is inserted in an electrical feedback
loop for said transmission channel.
17. A hearing aid comprising at least two input microphone channels, means
for adjusting the amplitude and phase characteristics of each of said
microphone channels, means for summing the outputs of said microphone
channels, an output receiver, a signal transmission channel connected to
receive the output of said summing means and to provide an output to said
receiver, and a programmable filter interposed between said summing means
and the output of said transmission channel, said filter being programmed
to impart to the hearing aid at least one response characteristic
effective to compensate for impaired hearing of the wearer of the aid and
to reduce the effects of both noise and reverberation.
18. A hearing aid comprising at least one input microphone, an output
receiver, and a signal transmission channel interposed between said
microphone and said receiver in which the improvement comprises a
signal-level dependent amplifier interposed between the input and output
of said transmission channel in a feedback path for said transmission
channel, said amplifier being programmed to impart to the hearing aid at
least one response characteristic effective to compensate for impaired
hearing of the wearer of the aid.
19. A hearing aid comprising at least one input microphone, an output
receiver, a signal transmission channel interposed between said microphone
and said receiver, and a programmable delay line filter interposed in a
feedback path between the input and output of said transmission channel,
said programmable filter being programmed to effect substantial reduction
of acoustic feedback from said receiver to said microphone. |
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Claims  |
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Description  |
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This invention relates to hearing aids, and more particularly to hearing
aids that are programmable so as to have suitable characteristics to
compensate for the hearing deficiencies of a patient. More specifically,
it relates to hearing aids of this character that are capable of
automatically adjusting to optimum parameter values as operating
conditions such as speech level, room reverberation and background noise
change, and also for reducing acoustic feedback.
BACKGROUND OF THE INVENTION
Conventional hearing aids suffer from several shortcomings. It is difficult
if not impossible with conventional hearing aids to provide a
frequency-gain characteristic that is ideal for each individual user. The
acoustic coupling between the hearing aid receiver and the eardrum also
introduces changes in the frequency-gain characteristic that is usually
deleterious to both speech intelligibility and overall sound quality. For
many patients, the optimum frequency-gain characteristic varies as a
function of the level of the speech signal reaching the hearing aid. In
order to protect patients from uncomfortably or dangerously loud signals,
it is also necessary to limit the maximum acoustic power output of the
hearing aid in some way. The methods used to limit acoustic power output
of hearing aids typically introduce deleterious distortions to the
amplified speech signal.
Another common problem is that of acoustic feedback. Even in the best
designed hearing aids, not all of the amplified acoustic signal is
delivered to the eardrum. A small proportion of the amplified acoustic
signal leaks back to the hearing aid microphone forming an acoustic
feedback loop. If the gain of the hearing aid is sufficiently high, this
acoustic feedback will cause a self-generating oscillation to occur,
resulting in an unwanted and highly unpleasant whistling sound. These
acoustic oscillations prevent the hearing aid from being used. Methods of
acoustic feedback control that are typically used include a tighter
acoustic seal between the earmold and the walls of the ear canal so as to
reduce acoustic leakage, placing the microphone at some distance from the
hearing aid receiver, e.g. on the opposite ear, or simply reducing the
gain of the hearing aid. None of these methods provides a satisfactory
solution for high-gain hearing aids.
One of the most common complaints of hearing aid users is that background
noise is particularly damaging to the understanding of speech. Methods
currently used to reduce background noise in hearing aids employ filtering
techniques in which the frequency regions containing high noise levels are
eliminated.
Another common problem is that of room reverberation produced by acoustic
reflections off the walls, ceiling, floor, and other surfaces in a room. A
small amount of reverberation is beneficial but too much reverberation
will make a room sound hollow or echoic and will interfere with both the
quality and the intelligibility of speech.
Systems have been proposed heretofore utilizing computers for testing the
hearing of patients and generating programming for a programmable hearing
aid, as disclosed in "Computer Application in Audiology and Rehabilitation
of the Hearing Impaired" by Harry Levitt, Journal of Communication
Disorders 13 (1980), pages 471-481, and in the U.S. Pat. Nos. 4,187,413 to
Moser, 4,489,610 to Slavin, and 4,548,082 to Engebretson et al., for
example. None of these systems, however, affords a satisfactory solution
for the problems of acoustic feedback, background noise, room
reverberation and changes in the optimum frequency-gain characteristic
resulting from variations in the level of the speech signal reaching the
hearing aid.
It is an object of the invention, accordingly, to provide a new and
improved hearing aid system that is free of the above-noted deficiencies
of the prior art.
Another object of the invention is to provide new and improved hearing aid
apparatus of the above character which is capable of automatically
adjusting to an optimum set of parameter values as the speech level and
type of background noise change.
A further object of the invention is to provide new and improved hearing
aid apparatus of the above character in which acoustic feedback is
substantially reduced.
Still another object of the invention is to provide new and improved
hearing aid apparatus that is capable of effective noise and reverberation
suppression and acoustic feedback reduction while maintaining optimum
hearing characteristics as the speech and noise levels vary.
SUMMARY OF THE INVENTION
A hearing aid system according to the invention comprises a hearing aid
that is programmable so as to have optimum electro-acoustic
characteristics for the patient and acoustic environment in which it is
used. It also includes instrumentation for measuring relevant audiological
characteristics of the patient, as well as techniques and instrumentation
for programming the hearing aid to have selected characteristics to
compensate for hearing deficiencies determined from the measurements made.
Desirably, several sets of optimum hearing aid parameter values, specified
in terms of both the amplitude and phase characteristics, are determined
for the patient as a function of speech level, type of background noise,
and room reverberation, both spectral and temporal characteristics of the
noise being taken into account. The selected optimum parameter values are
preferably programmed into an electronically erasable, programmable read
only memory (EEPROM) which supplies coefficients to programmable filter
and amplitude limiting means in the hearing aid so as to cause the hearing
aid to adjust automatically to the optimum set of parameter values for the
speech level, room reverberation, and type of background noise then
obtaining.
In one form, the programmable filter may be a digital equivalent of a
tapped delay line in which each delayed sample is multiplied by a
weighting coefficient. The sum of the weighted samples generates the
desired electroacoustic characteristics. Alternatively, the programmable
filter may be a tapped analog delay line in which the sum of the weighted
outputs of the taps generates the desired characteristics. An important
advantage of the latter type of filter is that the power consumption is
low and quasi-digital techniques can be used, i.e., the waveform is
sampled at discrete intervals in time without analog-to-digital
conversion.
Another form of filter uses a small number of delays in which the delayed
output is multiplied by a coefficient and added to the filter input so as
to achieve additional delays, a technique known as recursive filtering.
The invention also provides means for reducing acoustical feedback. In one
embodiment, an electrical feedback path in the hearing aid is matched in
both amplitude and phase to the acoustic feedback path and the two
feedback signals are subtracted so as to cancel each other. In an
alternative embodiment, a single filter in the forward path is used with a
transmission characteristic equivalent to that of the filter in the
forward path plus the electrical feedback path.
Environmental noise control is effected according to the invention by
providing means for sensing the relative speech/noise content in the
signals from the hearing aid microphone and generating binary words that
are supplied to the programmable filter for selecting from a memory a set
of delay line tap coefficients that are effective to impart to the filter
the appropriate frequency response for the specific environmental noise
condition then being detected.
According to the invention, reduction in both noise and reverberation is
achieved by the use of two or more microphones. The output of each
microphone is processed in both amplitude and phase such that the summed
output of the microphones is analogous to the output of a frequency
selective directional microphone.
DESCRIPTION OF THE PREFERRED EMBODIMENTS
For a better understanding of the invention, reference is made to the
following detailed description of a representative embodiment taken in
conjunction with the accompanying drawings, in which:
FIG. 1 illustrates schematically a host controller for use in prescribing a
wearable, programmable hearing aid according to the invention;
FIG. 2 illustrates schematically one form of programmable hearing aid
according to the invention which utilizes a digital delay line filter;
FIG. 3 shows a set of optimum frequency-gain characteristics appropriate
for different speech levels for a typical hearing impaired subject;
FIG. 4 is a partial schematic diagram of the hearing aid shown in FIG. 2,
modified to utilize an analog delay line filter; and
FIG. 5 shows schematically how multiple microphones can be used according
to the invention to reduce both noise and reverberation.
A hearing aid system according to the invention comprises generally a
wearable, programmable hearing aid in which all operations are controlled
by data stored in an erasable electrical programmable read only memory
(EEPROM), and a host controller providing electrical signals and test
sounds as necessary for measuring the residual hearing of a subject,
establishing optimal hearing aid parameters for the subject (including the
phase relationship between input and output) and generating control
signals as necessary to program the EEPROM module to perform the desired
operations in the hearing aid.
THE HOST CONTROLLER
Referring first to FIG. 1, the host controller 20 is shown as comprising an
audiometric signal generator 21 including a signal generator 22, the
frequency of which is controllable by a digital-to-analog (D/A) converter
23 controlled by a computer 24 through a latch 25. The output level of the
signal generator is adapted to be adjusted by an attenuator 26 also
controlled by the computer 24, through a latch 27. The output of the
attenuator 26 is supplied to one terminal 28 of a switch 29, from which it
can be transmitted through an amplifier 119 and a switch 120 to a terminal
connector 121.
The host controller 20 also includes a phase measurement circuit comprising
a digital phase shifter 30 which is adapted to receive the output voltage
of the hearing aid through a connector 31 and a programmable gain
amplifier 32, the gain of which is controllable by the computer 24 through
the latch 33.
In adjusting a hearing aid for reduced feedback as described below, the
input voltage developed by the hearing aid microphone in response to
acoustic feedback alone when a test signal is supplied to the hearing aid
is summed with an electrical feedback voltage supplied to the terminal 144
by the phase shifter 30. The summed acoustic and electrical feedback
voltages from the hearing aid are supplied from the terminal 34 through
the engaged movable and fixed contacts 35 and 36 of a switch 37, a
programmable gain amplifier 38, a rectifier 39 and an analog to digital
(A/D) converter 40 to the computer 24. In effecting the adjustment, the
gain of the amplifier 32 and the phase shifter 30 are adjusted until a
null output from the A/D converter 40 is read by the computer 24,
indicating cancellation of the electrical and acoustic signals. The
settings of the phase shifter 30 and the amplifier are then used to
program an EEPROM in the hearing aid so as to cancel acoustic feedback.
The programmable gain amplifier 38 is controlled by the computer 24 through
a latch 41, and the digital phase shifter 30 is also controlled by the
computer 24 through the latches 42 and 43.
Additional components of the host controller 20 are programming logic
comprising the latches 44 and 45 and a conventional tri-state buffer 46
which are activated in response to the computer 24 to provide the
necessary signals to program the EEPROM and control the hearing aid during
testing and programming, as described in greater detail below.
A conventional tape player 47 is also provided for generating various sound
combinations for testing of a patient's hearing. The tape player 47 is
controlled by the computer 24 through a latch 48 and its output level is
adapted to be controlled by an attenuator 49, also controlled by the
computer 24, through a latch 50. The output of the attenuator 49 is
supplied to the movable contact 51 of a switch 52 which is controllable to
supply the output either to a connector 53 to a sound field system, or to
a fixed contact 54 on the switch 29 for supply to the hearing aid, as
described in greater detail below. The switches 29 and 52 and the
tri-state buffer 46 are adapted to be controlled by the computer 24
through the latch 44.
The respective latches 27, 25, 42, 43, 33, 40, 41, 44, 45, 48 and 50 are
adapted to be controlled by the computer 24 through an address decoder 55
and Read/Write logic 56.
THE HEARING AID
Referring now to FIG. 2, a wearable hearing aid according to the invention
comprises a microphone 57, the output of which is fed through a
programmable automatic gain control (AGC) circuit 58 and a switch 59 to
one terminal of summing amplifier 60. In normal operation, amplifier 60
supplies the signal through the conductors 61 and 62 and a filter 63 to a
programmable filter 64 which is adapted to be programmed in the manner
described below to produce optimum hearing aid characteristics for the
patient based on the measurements made of the patient's residual hearing.
The output of the programmable filter 64 is fed from the movable contact
65 of the volume control 66, through a programmable limiter 67, and an
amplifier 68 to the hearing aid receiver 69.
The programmable filter 64 comprises essentially a digital tapped delay
line including a 12 bit A/D converter 70, which supplies outputs to a
32.times.12 random access memory (RAM) 71, a 12 bit temporary register 72,
and a 12 bit D/A converter 73. The signal supplied to the RAM 71 is
sampled by a 5 bit up-down counter 74 controlled by timing logic 75
connected to receive clock signals from an oscillator divider 76
oscillating at a frequency at least twice the audio signal band width.
The characteristics of the programmable filter 64 are determined by
coefficients stored in a random access memory (RAM) 77 which are selected
and fed to a 7 bit D/A converter 78, the output of which is supplied to
the D/A converter 73 for multiplication of the sampled data by a selected
coefficient from the RAM 77. The output of the D/A converter 73 is summed
in a conventional charge transfer summing circuit 78a and the sum signal
is supplied through a sample and hold circuit 79 to a conventional
anti-imaging filter 80. The output of the filter 80 is fed through the
movable contact 81 of a switch 83 and a fixed contact 82 thereof to the
volume control 66 and eventually to the receiver 69.
The coefficients stored in the RAM 77 when the hearing aid is in operation
are provided essentially by an EEPROM 84 previously programmed by the host
controller 20 (FIG. 1) as described in greater detail hereinafter. On
power-up, the filter coefficients and limit parameters are transferred
from the EEPROM 84 to the RAM 77 as follows: Tri-state switches 85 and 86
are enabled and power is supplied to the EEPROM 84 from a power-up control
circuit 87. A switch 88 is now activated to connect an 11 bit counter 89
to the RAM 77 through the switch movable contact 90. The power-up control
circuit 87 acting through the tri-state switch 85 supplies reset pulses to
a divide-by-12 counter 91, to the 11 bit counter 89 and to a series
parallel converter 92.
The power-up control circuit 87 also supplies clock pulses to the series
parallel converter 92 and to the divide-by-12 counter 91 such that after
every twelfth clock pulse, data is transferred from the EEPROM 84 through
the series parallel converter 92 and the tri-state switch 86 to one of the
memory locations in the RAM 77 as determined by the 11 bit counter 89.
These steps are repeated as many times as are required to transfer all of
the data stored in the EEPROM 84 to the RAM 77. When that occurs, the
tri-state switch 86 is disabled and supplies a signal to the switch 88 to
connect the movable contact 90 thereof to the fixed contact 93, which is
connected to the movable contact 94 of a switch 95 and to the 5 bit
counter 74 and the RAM 71. The hearing aid is then in its normal operating
mode.
Automatic adjustment of the hearing aid to take into account environmental
conditions such as changes in the speech level and type of background
noise obtaining at any time is effected by environmental control means
comprising a speech detector 96, four band pass filters 97, 98, 99 and
100, and a level detector including four differently prebiased comparators
101, 102, 103 and 104. Typical bandwidths for the filters 97, 98, 99 and
100 might be 100-750 Hz, 750-1500 Kz, 1500-3000 Hz, and 3000 to the upper
frequency limit, respectively. The speech detector 96 is conventional and
may include a level detector followed by a short term averaging device.
For steady state noise, the output of the level detector will be
relatively constant, indicating that noise only is present. Whenever
fluctuations in level are within a prescribed bandwidth typical of speech,
the short term average increases, indicating that speech is present. The
speech detector 96 is clocked periodically by signals from the oscillator
divider 76 to provide either a zero output indicative of noise or a one
output indicative of speech periodically to each of a plurality of sample
and hold circuits 105, 106, 107 and 108. The outputs of the band pass
filters 97, 98, 99 and 100 are rectified in the rectifiers 109, 110, 111
and 112, respectively and fed to the sample and hold circuits 105-108,
respectively, the outputs of which are fed to the comparators 113, 114,
115 and 116, respectively. The outputs of the rectifiers 109-112,
respectively, are also supplied directly as inputs to the comparators
113-116, respectively.
So long as there is a one output from the speech detector 96 indicating the
presence of speech in the input, the instantaneous outputs of each of the
band pass filters 97-100 are compared with previous values held in the
sample and hold circuits 105-108, causing the comparators 113-116,
respectively, to generate a binary coefficient (0,1) indicating whether or
not the speech level in the associated band pass filter exceeds the noise
level. At the same time, a level detector 117 responsive to the respective
outputs of the comparators 101-104 generates a two bit coefficient
indicating the average signal level.
The outputs of the comparators 113-116, inclusive, and of the input level
detector 117 are fed to latching means 118, which provides a six bit
output that is adapted to be transmitted through the switches 95 and 88 to
the RAM 77. Whenever the output of the speech detector 96 indicates that
speech is present, the binary outputs of the four filter level comparators
113-116, inclusive, and that of the average level detector 117 are
transmitted to the RAM 77 through the control switches 95 and 88. These
binary words are updated at regular intervals at a clock rate determined
by the oscillator divider 76, as stated. Each of the sixty-four possible
combinations of the 6 bit binary words identifies a different frequency
response for the programmable filter, and a corresponding set of
coefficients stored in the RAM 77 is selected, thereby automatically
adjusting the hearing aid to the optimum set of parameter values as the
speech level and type of background noise change.
In prescribing hearing aids with the use of the host controller, the
subject is seated in a quiet room with the hearing aid inserted in his
ear. The hearing aid is connected to the host controller by an electrical
cable (not shown), thereby placing it directly under the control of the
host controller. The prescriptive procedure usually consists of five
stages:
(1) measurement of the subject's residual hearing;
(2) derivation of an appropriate set of electroacoustic characteristics of
the hearing aid from such measurements, and programming the hearing aid
accordingly;
(3) measuring acoustic feedback in the hearing aid;
(4) re-programming the hearing aid to minimize acoustic feedback; and
(5) paired comparison testing of possible alternative settings of the
hearing aid to determine the optimal hearing aid settings for the subject.
The subject's residual hearing is measured using signals generated by the
audiometric section 21 in the host controller 20 (FIG. 1). These signals
are delivered to the hearing aid via the switch 29, an amplifier 119, a
switch 120, a connector 121, a connector 122 in the hearing aid, and the
conductor 62. The signals are processed by the digital programming filter
and associated circuitry in the hearing aid and are delivered to the
subject's ear using the hearing aid receiver and associated coupling that
the subject will actually wear after the hearing aid is prescribed. This
procedure eliminates the need for any corrections in going from headphone
to sound field measurements. The measurements are usually obtained with
narrow band stimuli (tones, warble tones, or narrow band noise) and
include threshold of hearing, various loudness levels (comfortable, loud)
and loudness discomfort level.
The measurements obtained on the patient's residual hearing are used in
deriving the electroacoustic characteristics of the hearing aid. The
measurements of loudness discomfort level are used to program the limiter
67 so that sounds amplified by the hearing aid never exceed the patient's
loudness discomfort level. The measurements of auditory threshold, most
comfortable loudness level, and loudness discomfort level are used to
determine the frequency gain characteristics of the hearing aid.
FIG. 3 shows four frequency-gain characteristics for a typical patient.
Curve A is used when the speech signal reaching the hearing aid is
relatively low, as would occur when somebody speaks in a very soft voice.
Under these conditions the hearing aid provides a large amount of gain,
particularly in the high frequencies. This is done to ensure that the
speech spectrum is placed above the patient's threshold of hearing at all
frequencies.
Curve B is used when the incoming speech signal is at the low end of the
comfortable loudness range for a normal hearing person. The amplification
provided places the speech spectrum at the bottom of the patient's most
comfortable loudness range at all frequencies.
Curve C is used when the level of the incoming speech is moderately loud
for a normal hearing person. The amplification provided places the speech
spectrum at the top of the patient's most comfortable loudness range for
all but the lowest frequencies. Less gain is provided at the low
frequencies to reduce upward spread of masking effects; e.g., weak
high-frequency sounds being masked by intense low frequency sounds.
Curve D is used when the signals reaching the hearing aid are very loud for
a normal hearing person. Under these conditions the hearing aid provides
relatively little gain with a significant roll off in the low frequency
region in order to substantially reduce upward spread of masking effects.
A set of coefficients is derived for each of these frequency gain
characteristics. These coefficients are derived using procedures that are
well known in the field of digital signal filtering and are used to
program filter 64 so that the hearing aid produces the required
frequency-gain characteristic. The filter coefficients are stored in the
RAM 77.
The level of the incoming speech signal is determined by the level
detectors 101, 102, 103 and 104. The decoder 117 generates a binary word
depending on the outputs of these level detectors. This binary word is
transmitted to the RAM 77, in order to select the appropriate set of
filter coefficients. If the signal reaching the hearing aid consists of
speech plus noise, as determined by the speech detector 96, then
alternative frequency-gain characteristics are used. These frequency-gain
characteristics are derived by first determining the incoming signal
level, as described above, selecting an appropriate frequency-gain
characteristic and then reducing the gain in those frequency regions where
the background noise level exceeds the speech level. This is determined by
comparing the output levels of the bandpass filters 97, 98, 99, and 100
when speech is present to the corresponding levels measured when noise
only is present. The latter information is stored in the sample and hold
units 105, 106, 107 and 108. The comparisons are done by means of the
comparators 113, 114, 115, and 116. The outputs of these comparators in
combination with the outputs of the level decoder, 117, generate a 6-bit
word that selects the appropriate set of filter coefficients in the RAM
77.
For patient hearing parameter selection and programming, the hearing aid is
interfaced with the host controller as described above, and the EEPROM 84
(FIG. 2) is plugged into a programming slot 124 in the host controller 20.
A conductor in the line 125 (FIG. 1) is set to logic 1 by operation of the
computer 24 which applies a logic 1 signal to the line 126 (FIG. 2),
resulting in the tri-state switch 85 being disabled and the switch 88
being activated by an OR gate 127 to move the switch movable contact 90 to
connect the counter 89 to the RAM 77. The host controller 20 generates
reset pulses which are transmitted over a conductor in the line 128 (FIG.
1) through the connector 129 and the line 130 in the hearing aid (FIG. 2)
to reset the counters 91 and 89 and the series-to-parallel converter 92.
Clock signals are fed from the host controller 20 over another conductor in
the line 128 through a connector 131 and a conductor 132 in the hearing
aid to the serial-to-parallel converter 92 and are divided by 12 in the
counter 91, the output of which is fed through the switch 88 to the RAM
77. Synchronously with the clock signals on the line 132, data are fed
from a conductor in the line 128 in the host controller 20 and through a
connector 133 and a conductor 134 in the hearing aid to the
serial-to-parallel converter 92. After every twelfth clock signal, data is
transferred from the serial-to-parallel converter 92 through the tri-state
switch 86 to one of the memory locations in the RAM 77 determined by the
counter 89. This step may be repeated as many times as required to store
essential data in the RAM 77 within the storage limits of the latter. The
line 126 is then set to logic 0 by operation of the computer 24 through a
conductor in the line 125, thus disabling the tristate switch 86 and
actuating the switch 88 to cause the movable contact 90 thereof to move
into engagement with the fixed contact 93 resulting in the connection of
the RAM 77 to the counter 74 and to the RAM 71. The hearing aid is now
ready for patient hearing parameter selection and programming.
The selection of the desired hearing aid parameters is accomplished by
actuating the computer 24 to set the line 135 in the hearing aid (FIG. 2)
to logic 1 through a conductor in the cable 128 and a connector 136 on the
hearing aid. This activates the switch 95 and supplies data in the form of
6 bit words from the series-to-parallel converter 92 over the line 137
through the movable contact 94 of the switch 95 and the movable contact 90
of the switch 88 to the RAM 77. The series-to-parallel converter 92 is
reset by a signal from the computer 24 (FIG. 1) transmitted through a
conductor in the line 128 and the connector 129 and conductor 130 in the
hearing aid, and a 6 bit word is fed into the converter 92 and placed on
seven of the address lines of the RAM 77, selecting one of sixty-four
possible sets of coefficients for use in the programmable filter in the
hearing aid. The selection proceeds in the like fashion throughout the
process of frequency shaping of the filter.
Feedback cancellation is achieved in the hearing aid by first measuring the
acoustic feed back in situ and then creating an electronic feedback path
with identical amplitude and phase characteristics. The outputs of the two
feedback paths are then subtracted, thereby canceling any feedback signals
that might occur.
Acoustic feedback is measured with the hearing aid in the ear as it would
normally be worn. An electrical test signal is applied to the terminal 122
(FIG. 2) from the host controller 20. A portion of this amplified acoustic
signal will leak through the ear mold and reach the microphone 57, which
will then convert the signal back to electrical form and return it to the
hearing aid amplifier. Feedback will occur if the total gain in the loop,
i.e., from the input to the filter 63 through the filter and amplifier of
the hearing aid to the output transducer 69 and from the microphone 57
back to the input to the filter 63, exceeds unity.
For the purpose of this measurement, the feedback loop is broken between
140 and 122. Terminal 140 of the hearing aid is then connected to terminal
34 of the host controller, terminal 142 of the hearing aid is connected to
terminal 31 of the host controller, and terminal 143 of the hearing aid is
connected to terminal 144 of the host controller. The programmable phase
shifter 30 and programmable amplifier 32 are then adjusted by the computer
24 so as to minimize the sum of the acoustic and electrical feedback
signals of the output of summing amplifier 60.
From the settings obtained with t | | |