Note: Descriptions are shown in the official language in which they were submitted.
W092/0~330 ~ ~9 ~3~ PCT/AU9t/00506
BIMODAL SPEECH PROCESSOR
Technical Field
The present invention relates to improvements in the
proces~ing of sound for the purposes o~ supplying an
information sign~l to either an acoustic hearing aid, a
cochlear implant aid device or both so as to improve the
quality of hearing of a patient.
Back~round of the_Invention
Throughout this specification, reference to an acoustic
hearing aid is reference to an aid of the type adapted to fit
in or adjacent an ear of a patient and which provides an
acoustic output suitable to at least partially compensate for
; hearing deflclencies o~ the patient. Throughout th~s
~peci~ication a cochlear implant aid will refer to a devlce
' which i~clude~ components which are ~itted wlthin the body of
a patient and which are adapted to electrically stimulate the
nervous system of a patient in order to at least partially
compensate for usually pro~ound hearing loss of the patient.
There is a trend towards fitting cochlear implants to
patients with some residual hearing in the contralateral ear.
; Many patients recognise speech better using conventional
acoustlc hearing aids together with the cochlear implant than
they do using either device alone but ~ind the use of the
; combinatlon unacceptable. These patients opt to use either
the acoustic hearing aid or the cochlear implant aid but not
both devices toge~her.
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W092/08330 PCT/AU91/00506
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It is an object of the present invention to provide a
: bimodal aid device which can drive both an acoustic hearing
aid and a cochlear implant aid which thereby improves the
qual~ty of binaural information received by a patient.
Two further problems experienced iIl the prior art of
~earing aids are (l) quickly and easily measuring the nature
and degree of hearing impairment of a client for the purposes
of providing an appropriate hearing aid and (2) the
difficulty in matching appropriate heariny aid qualities and
~0 capabilities to the qpecific requirements of the user
Recently, a few hearing aid devices have appeared on the
~ar~et which allow on-line control of ~he gain
characteristics of the device at different frequencies.
However, these devices do not provide speech processing
lS capability such a~ is provided by formant extraction and like
feature extraction circuits.
It is a further ob~ect of particular e~bodiments~of the
pre~ent invention to provide a signal proce~sing device for
use in association with an acoustic hearing aid which
addresses these problems.
SummarY of the Invention
Accordingly, in one broad form of the invention there
is provided a bimodal aid for the hearing impaired which
includes processing means adapted to receive and proce~s
: 25 audio information received from a microphone; said processing
~eans supplying processed information derived from ~aid audio
information to an implant aid adapted ~o be implanted in a
first ear of a patient and to an acoustic aid adapted to be
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W09~/0833~ ~o~-3~ PCT/~U91/0~5~6
worn in or adjacent a second ear o~ said pa~ient wnereby
binaural information is provlded to said patient.
In yet a further broad ~orm of the invention there is
provided a bimodal aid for the hearing i~mpaired comprising a
sound/speech proce~sor electrically connected to a hearing
aid transducer adapt~d to be worn adjacent to or in an ear of
a patient and electrically connected to an electrical signal
: transducer adapted to be worn in an ear of a pa~ient, ~aid
: speech processor r~ceiving and proce~sing audio input
information so as to produce an acoustic si~nal from said
hearing aid transducer and an electrical i~nal from said
electrical transducer whereby coherent blnaural information
ls provided to said patlent.
In a further broad form there is provided an
electronically co~figurable sound/s~eech processor for the
hearlng impaired; said processor including configuratio~
means an~ signal processing means; said proce~sor adapted to
receive audio information and to process said audio
: informatlon by said signal proce~sing means in accordance
with parameters set by said configuration ~eans ~o as to
produce an output signal adapted to stimulate a hearing aid
~ransducer; said configuration means adapted to receive one
or more of electronic signal input or software input for the
purpose of modifying said parameters.
In a further broad form of the invention there i~
. provided 2 method of and a means for control of a hearing aid
and a cochlear implant by means of said sound/speech
proce~sor.
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In a particular form said sound/speech processor
utilises the speech features FO, Fl, F2, AO, Al, A2, A3, A4,
A5 and voiced/voiceless sound decisions to produce said
output signal adapted to stimulate a hearing aid transducer,
wherein said features are defined as follows:-
FO is the fundamental frequency,
.. Fl is the first formant frequency,
F2 is the second formant frequency,
` AO is the amplitude of FO,
. Al.is the-amplitude of Fl,
A2 is the amplitude of F2,
A3 is the amplitude of band 3: 2000 to 2800 Hz,
A4 is the amplltude o~ band 4: 2800 to 4000 Hz,
A5 is the amplltude of band 5: ~000 Hz and above.
In a further partlcular form of said ~ound/speech
,, processor, 3aid signal processi~g means includes means for
dynamically changing the amplitude of different frequency
bands in the defined speech spectrum as a ~unction o~ the
-.~, speech features Al, A2, A3, A4, and A5 parameters so that the
loudness in these bands is appropriately scaled between the
. threshold and maximum comfortable levels of the hearing aid
' user at the Fl and F2 frequencies and in the higher frequency
`i bands.
In a further particular for~ of said sound/speech
proce~sor, said signal processing means includes filter means
whose settings are dynamically varied according to speech
parame~ers extracted from said signal processing means
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W092/08330 2 ~ ~ a 3 4 4 PCT/AU91/00506
whereby said filter means dynamically adapts said output
signal to overcome the effects of noise and/or particular
deficiencies in the hearing of a user.
In yet a further particular form of said sound/speech
processor, said signal processing means w:ithin said processor
includes means for reconstructing signals in real time
whereby the amplitude and/or frequency characteristics of
said output signal can be controlled optimally so as to
enhance speech recognition in a user.
In a first mode of operation of said sound~speech
processor, said filter means set by said conflguration means
are prov~ed ba3ed on measurements set by an audiologist
during a hearlng aid ~lttlng procedure. The filter settings
remaln fixed aSter completion of the procedure and remain
flxed thereafter.
In an alternatlve ~ode oP opera~lon of said
~ound/speech processor, said proce~qor includes ~ilter means
whose parameters are changed dynamically while sald processor
is in use providing said output signal to said hearing aid
transducer in accordance with informatlon provided to said
configuration means by speech parameter extraction means
acting on said audio information.
In a further particular mode of opera~ion of said
sound/speech processor said output signal is synthesised by
said signal processing means utllising only speech
para~eters.
Brief Description of the Drawin~s
E~bodiments of the invention will now be described with
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W092/08330 PCT/AU91/00~06
~ )o9-~344 -6- `
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reference to the drawings wherein:-
Fig. l is a schematic representation of a bimodal
aid according to a first embodiment of the
invention,
Fig. 2 is a schematic representation of the main
functional components which comprise the entire
bimodal aid,
Fig. 3 is a schematic block diagram o~ the implant
processing circuitry together with the acoustic
-processor circuitry, ~ ~ ~~~~
Fig. 4 is a chart showing an example of the pattern of
electrical stimulation of the implant
electrodes for varlous steady state phonemes
using the multi-peak codlng strategy,
Flg. 5 is a graph showlng the standard loudness growth
function for the speech proce~sor portion when
. ' driv~ng the implant,
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Fig. 6 is a schematic block diagram o~ the functional
components of thc bi~odal proces~or which drive
the acoustic aid,
Fig. ~ is a schematic block diagram of the components
comprising the acoustic proc~ssor which
proces~es the speech signal in accordance with
" particular forms of the invention to drive the
acoustic aid,
:
Fig. 8 is a component block diagram of the acoustic
processor,
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W092/0$330 ~ 4~ PCT/AU91/00506
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. Fig. 9 is a component block diagram of a Biquad filter
as utllised in the acoustic processor of Fig.
8,
Fig. 10 is a component block diagram of the output
` 5 driver ~or the acoustic aid,
- Fig. 11 is a schematic representation o~ pre~erred
modes of operation of the acoustic proces~or
- for voiced vowel input,
~ig. 12 shows a sound intensity again~t ~requency for
an acou~ic aid operatlng accordin~ to mode-1-; - ~
Fig. 13 shows comparative plots of sound in~enslty
against frequency in relation to an example of
mode~ one and three operatlon o~ the acoustic
proce~sor,
Fig. 14 outllne~ a fltting ~trategy ~or the blmodal
aid,
Flg. 15 i~ a ~lowchart o~ a fittin~ ~rategy for the
; acoustlc aid portion of the device iss mode 1,
; and
Fig. 16 i a schematic block diagram of the ut~lisation
: of the diagnostic and programming unit for use
in association with the bimodal aid.
- ~escri~tiosl of Pre~erred ~mbodi~ents
With reference to ~ig. 1, the bimodal aid i5 a hearing
aid device which has the ~apability to provide issformation
through a cochlear implant aid in one ear and a speech
processing acoustic hearing aid in the other ear o~ a
patlent. Both the implant aid and the acoustic aid are
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W092/08330 PCT/AU91/005~
2~9~34~ -8-
controlled by the same speech processor which derives its raw
audio input from a single microphone.
The speech processor extracts certain parameters from
the incoming acoustic wavefor~ that a~e relevant to the
perception of speech sounds. Some of the! speech parameters
extracted by the speech processor are translated in~o an
electrical signal and delivered to a cochlear implant. These
; features can also be u~ed as a basis for modification of the
speech waveform following which it is then amplified and
presented to-the acoustic aid. - ~
There are some patlents with a small amount of residual
hearing who have already received an implant, and who have
previously worn hearlng aids in the non-implanted ear. These
patients often report that the sounds produced by
conventional acoustic hearlng alds are incompa~lble with
those produced by the implant. Such pa~ients tend ~o resort
to one or the other and thu~ do not make maximal use of their
limited auditory capacities. These patients are candidates
for the bimodal aid. Such a device incorporating a cochlear
implant aid and a speech proce~slng acous~ic aid can provide
information which will allow these patients to discrimlnate
speech better than any currently available hearing dev.tce
alone.
; Generally, if pa~ients have ~ome residual hearing, it
tends to be low frequency. The cochlear implant produces
stimulation at positions ln the cochlear that correspond to
higher frequencies ~u~ually above ~OOHz). Thus, by combining
the two channels it is possible to provide useful information
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W092/08330 ~ Q~9 ~ 3 ~ ~ PCT/AU9t/00506
_g_
over a much wider range of frequencies than either channel
could provide alone. Furthermore, the frequency and temporal
resolution of residual hearing can be better than that
provided by the pulsatile electric signal of the cochlear
implant aid portion o~ the bimodal aid.
In addition to the above "bimodal" uses the acoustic
aid driver part of the device can also be used as a speech
processing hearing aid independent of the cochlear implant
aid output. When used in thls manner it has advantages over
conventional acoustic hearing aids. Conventional hearing
aids are limited in practice because the adjustments to the
frequency/gain characterlstics are restricted to a small
,~ number of options and there are many users who are not
, optimally alded. There is a need for a hearing aid with a
;, 15 more ~lexlble frequency/gain characteristlc and this can be
achieved with this aid. In addition, the feature extraction
clrcuits which are the basis of the cochlear implant aid
allow the hardware tu measure important characteristics of
th0 speech signal in quiet conditions and in conditions of
moderate amounts of background noise. These characteristics
can then be amplified selectively and enhanced relative to
the rest of the acoustic signal, or used to synthesize a new
speech-like wa~eform that carries the sa~e information
exclusively. This is performed by the acoustic signal
processor (12) which outputs to an acoustic aid.
The synthesized waveform is used to overcome special
problems. For example, high frequency sounds above the limit
of a user' 5 hearing can be presented as lower frequencies
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W092/08330 ~ 0 3 ~ 3 ~ 4 - 1 o- PCT/AU91/00506
within the user's hearing range. Broad peaks in the speech
spectrum can be made narrower if this provides better
frequency resolution and helps to reduce masking of one peak
by other adjacent peaks. There is no other single, wearable
device capable of implementing all these processes.
The sound/speech proces-~or can take in a speech ~gnal
from a microphone, measure selected features of that signal
(including the frequency and amplitude of formants for voiced
speech) and control the outputs to both a cochlear implant
aid and an acoustlc hearing aid in the ca~e of the first ~ ~~~~
embodiment or only the acoustic hearing aid in the case of
the second embodiment.
l.O FIRST ~MBODIMENT - BIMODAL AID
Figure 2 shows a schematic diagram of the op~ratlon of
the device. The cochlear implant aid portlon of the dev1ce
i~ covered by existing patents or patent applications to the
same applicant and the implant aid operates upon one ear of a
patient using similar strategies to tho~e already developed
for implant users. In addition, the users of the bimodal
device will receive an auditory signal via an acoustlc aid in
the non-implanted ear. The capabilities of the bimodal aid
allow this signal to be specially tailored in order to convey
information co~pl0mentary to the implant and utilise the
residual hearing of the patient maximally.
Specifically, Fig. 2 discloses the body worn portion of
the bimodal device comprises a speech proce~or ll intimately
connected to an acoustic aid processor 12 together with a
microphone 13, an acoustic hearing aid 14 and an implant aid
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W092/08330 2 ~ 9 ~ 3 ~ '1 PCT/AU91/00506
15. The implant aid ;5 comprises an electrode array 16
electrically connected by harness 1~ to a receiver stimulator
18 which is in radio communication with speech processor 11
by way of internal coil 19 and external coil 20.
In addition Fig. 2 shows auxiliary ite~s being the
diagnostic and programming unit 21 and the diagno~tlc
programming interface 22.
Currently the diagnostic and programming u~it 21 is
implemented as a program running on a personal computer
; 10 .whilst the diagnostic programming inter~ace 22 is a - ---~~ ~ ~~
communlcatlons card connected to the PC bus. The diagnostic
and progra~ming unit 21 is utilised in a clinical situation
to test ~or and control device paramet~rs of operatton ~or
the ~peech processor 11 and/or acoustic aid proces~or 12
whlch optlmi~e hearlng performance for a patient according to
de~lned crlteria. The~e parameters are communicated via the
dia~nostlc program~ing lnterface 22 to a ~ap memory st`orage
23 in the ~peech processor 11. It i5 by reference to ~he
parameters stored in the map memory ~torage 23 that the
manner of proce~sing of the audio signal received from
microphone 13 is deter~ined both for the speech processor 11
when driving the implant aid 15 and the acoustic aid
processor 12 when driving the ~coustic ald 14.
The co~ponents illustrated in Fig. 2 other than the
acoustic aid processor 12 and the acoustic aid 14 and the
computer program controlling the function of ~he diagnos~ic
and programming uni~ 21 have been described el~ewhere in
earlier f iled patents and patent applica~ions and remain the
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W092~08330 2 ~ 9 ~ ~ 4 4 PC~/AU9ttO0506
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same in so far as operation of the Cochlear implant aid is
concerned.
The speech processor 11 and the preci~e methodology for
exciting electrically the implant aid has varied since the
- 5 inception of these devices and can be expected to continue to
vary. For example excitation of the sti~nulating electrodes
placed within the ear of a patient can be either digital or
analogue in nature. TQ date, one of the present applicants
Cochlear Pty. Li~ited, has pursued a strategy of digital
electronic stimulation using what have been termed pulsatile
electrical signals applied to a pulsatile electrical signal
transducer.
Particularly, the speech processor 11 has been
commercially available in a numb~r of form~ since around 1982
from Cochlear Pty. Llmited (one of the co-appllcants for the
present application~. The early units and, indeed, even the
most'r2ce~t unlts are primarily aimed at improving speech
perception in favour of all other soundR received from
:~ microphone 13. This is done by causlng speech processor 11
to di~cern and proce~s from the raw audio lnput received from
microphone ~3 acoustic features of speech which have been
determined to best characterlse speech information as
perceived by the human auditory system.
;:
` ~arly forms of the speech processor 11 presented three
:~ 25 acoustic fea~ures of speech to implant users. These were
. amplitude, presented as current level of electrical
- stimulatio~; funda~ental fre~uency ~FO) or voice pitch,
presented as rate of pulsatile stimulation; and the second
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W092/08330 2 ~ 9 3 4 4 PCT/AU91/00506
-13-
formant frequency ~F2) represented by the position of the
stimulation electrode pair located within the ear of the
patient. The F2 frequency is usually found within the
frequency ranse 800 to 2500 Hz.
Later a second stimulating electrode pair was added
representing the first formant (F1) of speech. The F1 signal
is typically found within the frequency ranye 280 ~z to 1000
Hz. This scheme ~known as the FOFlF2 schemel provided
improved performance in areas of speech perception as against
the earlier FOF2 scheme~ In most recent times--the-
information provided to and processed by the speech processor
11 has been increased with one particular purpose being to
improve speech intelligibility under moderate level3 of
background nolse.
This latest coding scheme provides all of the
information available in the FOFlF2 scheme while providing
additional information from three high frequency band pass
filters. These filters cover the followin~ frequency ranges:
2000 to 2800 Hz, 2800 to 4000 Hz and 4000 to 8000 Hz. The
- 20 energy within these ranges controls the amplitude of
electrical stimulation of three fixed electrode pairs in the
basal end of the electrode array. Thus, additional
information about high frequency sounds is presented at a
tonotopically appropriate place within the cochlear.
The overall stimulation rate for voiced sounds remains
as FO (fundamental ~requency or voice pitch~ but in the new
scheme four electrical stimulation pulses occur for each
W092/08330 PCT/AU9l/00506
2 ~ 4~ -14-
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glottal pulseO This compares with the FOFlF2 strategy in
which only two pul es occur per voice pitch period. In the
new coding scheme, for voiced speech sounds, the two pulse5
representing the first and second formant are still provided
and additional stimulation pulses occur representing energy
in the 2000 to 2800 ~z and the 2800 to 4000 Hz ranges.
For unvoiced phonemes, yet another pulse representing
energy above 4000 Hz is provided while no stimulation for the
first formant is provided, since there is usually little
energy in this frequency range. Stimulation occurs at ~`~
random pulse rate o~ approximately 260 Hz which is about
double that used in the earlier strategy.
The latest noise suppression algorith~ operates in a
continuous manner, rather than as a volce activated switçh as
16 previously been used. This removes the perceptually annoyin~
switchlng on and of~ of the earlier system. In the new
algorlthm the noise flow is continuously asse~sed in each
frequency band over a period of ten seconds. The lowest
level over this period is assumed to be background noise and
is subtracted from the amplitude relevant to that frequency
band. Thus any increaRe in signal amplitude above the noise
level is presented to the patient while the ambient noise
; level itself i9 reduced to near threshold.
Fig. 3 illustrates the basic filter a~d prccessing
structure o~ a bimodal aid incorDorating means ~o implement
any o~ the above desoribed latest proc~sslng scheme.
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W092/08330 2 a ~ PCT/AU91/00506
-15-
International Patent Application PCT/AU90/0040~ to the
present applicant entitled Multi-peak Speech Processor
describes in detail the operation of the~e components. The
entire text and drawings of the specification of that
application are incorporated herein by cross reference. The
most pertlnent portions of that specification are included
immed~ately below.
The nature of the electrode array that is utilised in
con~unction w1th the latest coding strategy and the manner
-and nature of its implantation is described in the ~
literature, for example 6 chlear ~rosthe3es, editors Clark
.M., Tong G., Patrick; publlshed by Churchill Livingstone
1990. Chapter 9 of that book entitled "The Surgery of
Cochlear Implantation" by Webb R.L., Pyman B.C., Franz B. K-H
lS and Clark G.M. i8 partlcularly pertinent. The text and
drawings of that chapter are incorporated herei~ by cross
re~erence. I
The coding strategy extracts and codes the F1 and F2
spectral peakR from the microphone audio signal, using the
extracted frequency estimate~ to select a more apical and a
more basal pair of electrodes for stimulation. Each selected
electrode is stimulated at a pul~e rate equal to the
Pundamental fre~uency F0. In addition to F1 and ~2, three
high frequency bands of spectral information are extracted.
The amplitude estimates from band three ~2000-2~00 Hz), band
four (2800-4090 Hz), and band five (above 4000 Hz) are
prese~ted to fixed electrodes, for example the seventh,
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W092/08330 PCT/AU91/00506
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fourth and first electrodes, respectively, of the electrode
array 16 (Fig. 2 and Fig. 4).
- The first, fourth and seven~h electrodes are selected
~ as the default electrodes for the high-fre~uency bands
because they are spaced far enough apart 50 that most
patients will be able to discriminate between stimulation at
these three locations. Note that these derault assignments
may be reprogrammed as required. If the three high frequency
bands were asslgned only to the three most basal electrodes
in the..MAP,.many patients might not find the additional high
frequency information as useful since patients often do not
demonstrate good place-pltch discrimination between ad~acent
ba~al electrodes. Additionally, the overall pltch percept
.. resultlng from the electrical ctimulatlon might be too hlgh.
Table I below indlcate~ the frequency ra~g~s of ~he
various formants employed in the speech coding scheme for the
present ~nvent~on.
, TABLE I
Frequ~ncv Ran~e Formant or Band
20280 - 1000 Hz Fl
800 - 4000 Hz F2
2000 - 2800 Hz Band 3 - Electrode
2800 - 4000 Hz Band 4 - Electrode 1
4000 Hz and above Band 5 - Electrode 1
If the input signal is voiced~ it has a periodic
:` fundamental frequency. The electrode pairs selected from the
` estimates of Fl, F2 and bands 3 and 4 are stimulated
W092/08330 '~ ~ 9 j ~ 4 4 PCT/AU91/00506
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sequentially at the ra~e equal to F0. The most basal
electrode pair is stimulated first, followed by progressively
more apical electrode pairs, as shown in Fig. 4. Band 5 is
not presented in Fig. 4 because negliglble information is
contained in this frequency band for most voiced sounds.
If the input signal is unvoiced, ener~y in the Fl band
(280-lO00 Hz) is usually zero. Consequently it is replaced
with the frequency band that extracts information above ~000
Hz. In this situation, the electrodes pairs selected from
the estimates of F2, and bands 3, 4 and 5 receive the
pulsatile stimulation. The rate of stimulation is periodic
and varies between 200-300 Hz. The codin~ strategy thus may
be seen to extract and code five spectral peaks but only four
` spectral peaks are e~coded for any one stlmulus sequence.
FIG. 4 illustrates the pattern of electrlcal
: stimulation for various steady state phonemes when using this
coding~strategy. A primary function of the MAP is to
translate the frequency of the dominant spectral~peaks ~F1
and F2) to electrode selection. To per~orm this function,
the electrodes are nu~bered ~e~uentially starting at the
round window of the cochlea. Electrode 1 is the most basal
electrode and electrode 22 is the most apical in the
electrode array. Stimulation of di~ferent electrodes
normally results in pitch perceptions that reflect the
tonotopic organization of the cochlea. Electrode 22 elicits
the lowest place-pitch percept, or the "dullest" sound.
Electrode 1 elicits the highest place-pitch percept, or
"sharpest" sound.
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W092/08330 PCT/AU91/00506
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To allocate the frequency range for the Fl and F2
spectral peaks to the total number of electrodes, a default
mapping algorithm splits up the total number of electrodes
available to use into a ratio of approxirnately 1:2, as shown
in FIG. 4.
Inside the speech processor a random access memory
stores a set of number tables, referred to collectively as
the MAP memory storage 23. The MAP determines both stimulus
parameters for Fl, F2 and bands 3-5, and the amplltude
estimates. The encoding of the stimulus parameters follows a
equence of distinct steps. The steps may be summarized as
follows:
1. The first formant frequency (F1) is converted
t~ a number based on the dominant spectral peak in the region
between 280-1000 Hz.
2. The Fl number is used, in conjunction wlth one
of the MAP tables, to determine the electrode to be
stimulated to represent the first formant. The indifferent
electrode is determined by the mode.
~- 20 3. The second formant frequency (F2) i9 converted
to a number based on the dominant spectral peak in region
between 800-4000 Hz.
4. The F2 number i5 used, in conjunction ~ith one
of the MAP tables to determine the electrode to be stimulated
to represent the second formant. The indifferent electrode
is determined by the mode.
W092/08330 2 ~ ~} ~ 3 ~ ~ PCT/AU91/00506
--19--
5. The amplitude estimates for bands 3, 4 and 5
are assigned ~o the three default electrodes ?, 4 and 1 for
bands 3, 4 and 5, respectively, or such other electrodes ~hat
may be selected when the MAP is being prepared.
6. The amplitude of the acous-tic signal in each of
the frequency bands i8 converted to a number ranging from 0 -
150. The level of 3tlmulation that will be delivered is
- determlned by referring to set MAP tables tha~ relate
acoustlc amplitude (in range of 0-150) to stimulatlon level
for the-specific electrodes selected in steps 2, 4 and 5,~ ~~~~ ~ ~
above.
7. The da~a are further encoded in the speech
proce~sor and tran~mitted to the recelver/stimulator 18. It,
in turn, decodes the data and send~ the stlmull to the
appropriate electrodes. Sti~ulus pulses are pre~ented at a
: rate equal to F0 during voiced periods and at a random a
periodlc rate within the range of FO and F1 formants
(typically 200 to 300 Hz~ during unvoiced periods.
The speech processor 11 additionally includes a non-
linear loudness growth algori~hm that conver~s acoustic
~ignal amplitude to electrical stimulation parameters. The
speech proce~sor 11 converts the amplitude of the acoustic
signal into a d~gital linear scale with values ~rom 0 - 150
as shown in FIG. 5. That digital scale ~in combinatio~ wlth
the information stored in the patien~'s MAP) determines ~he
actual charge delivered to the electrodes in the electrode
array 16.
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Improvements on this assembly are disclosed in co-
pending applications to Cochlear Pty. Limited. Specifically
International Application PCT/AU90/00406 discloses an
improved connection system between mi~rophone 13 and speech
processor 11 and between the external coil assembly 20 and
the speech processor 11. The text and dra~ings o~ the
specification of that application are incorporated herein by
cross reference.
A noise suppression circuit is disclosed in
International Patent-Application PCT/AU90~00404. The text
and drawings of the specification which accompanied ~hat
, applicatio~ are incorporated herein by cross reference.
FIG. 6 is a block diagram of the processing circultry
showing the functional interconnection of components for
driving the acoustic heariny ald 14. The main components
comprise ~lcrophone 13, automatic gain control 24, speech
parameter extractor 25, encoder 26, patient MAP ~emory
storage 23, noi~e generator 2~ and acoustic aid si~nal
processor 12.
The heart of the bimodal aid as far as allowlng the
acoustic aid 14 to be driven from the speech processor 11 is
the acoustic aid slynal processor 12.
The acoustic aid signal processor is software
confi0uratlve and contains three two-pole filters each of
which can be used in either bandpass, lowpass or highpa s
configura~lon. The centre frequency, bandwidth and output
amplitude o~ these filters are controlled by the processor.
W092/08330 ~ PCT/AU91/00506
-21-
The filters can be used in series or in parallel and the
input waveform can be the speech waveform, pulses, a noise
signal or external signal. The external signal can be from
another microphone, other acoustic output or another
acoustic signal processor. This results in a particularly
flexible aid that can operate either in a ~anner similar to a
conventlonal acoustic hearing aid (though with more accurate
gain fitting than most currently available aids- can provide)
or as an aid providing different types of processed speech
10 information. The acoustic aid signal processor including-the -
three programmable filters has been implemented on a single
silicon chip. Each filter is usable as a high-pass, band-
pa~s, or low-pass filter. ~28 centre frequencies between
lOOHz and 16000 Hz, 128 Q value~ between 0.53 and 120, and
15 128 amplitude value~ between 0 and 64 dB are available for
each filter (Q=centre ~requency/B.W.). This chip has the
flexibility to cover a wide range of krequencies, amplitudes
; and spectral shapes.
It also includes a dlgital-to-analog converter (DAC)
20 that is used to produce the excitation waveform for the
filters. The DAC can produce waveforms of arbitrary shape
t9UCh as sinusoidal or pul3atile) controlled directly by the
processor, or can be switched to provide excitation by the
speech wave~orm, or a whlte noise generator.
A schematic dia~ram of the three-filter circuit is
shown in Fig 7.
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W092/08330 ,~ 9 3 ~4~ PCT/AU91/00506
: A functional specification for a single chip
. implementation of the acoustic aid signal processor 12 is
provided by FIGS. 8, 3 and 10. Details of the specification
are as follows: v
5 Topologv
Flg. 8 shows the overall topology of the chip. Three
programmable f~lters in which centre frequency and band width
can be independently controlled are provided. The outputs of
these filters can be independently attenuated or amplified
and then mixed.- -The output o~ one of the three filters can ~~~
be inverted if necessary by setting an INV bit.
Fig. 9 shows detalls of one of the Blquad filters
formlng the three filter array together wlth the ~requency
latches and Q latches which determlne the parameters of the
Biquad filter.
The topology of the chip can be altered from serial to
parallel or a mixed structure by three PARn bi~s.
The signal source for this structure can be selected by
a four channel multlplexer ~MUX). This ~elects ~5 volts, a
buffered outpu~ of the audio signal, an internally generated
noise source, or an external signal. This signal source is
: ~ed to a 7 bit digital to analog converter ~DAC) as a
reference voltage.
The multiplying DAC can convert the DC level into a
pulse generator, or provide a fine gain control on the audio
or external signal, or noise source . The most signiflcant
bit ~MSB) is used to invert the output.
W~92/08330 ~ ~ 9 ~ 3 ~ ~ PCT/~U9l/00506
-23-
All filter outputs are summed and passed to a push pull
earphone driver which can provide effectively lO volts peak-
to-peak across a 2~0 ohm (nominal) earphone. The chip uses a
single supply of 5 volts.
Note that the earphone has a DC re~sistance of 88 ohm
with the impedance ri ing gradually to 270 oh~ at 1 kffz. The
output stage consists of a bridge of P and N transistor
switches as shown in Fig. 10. The switches are pulse width
modulated by a signal derived ~rom a comparator driven from a
triangle wave~on one input and the aud1o signal on the other.
The on re~i~tance of the switches should be less ~han 5 ohms
~lower if possible).
Apart from the class D output, there iQ a slngle ended
linear output. This should be capable of sourcing or sinklng
5 mA with le~s than 1 volt drop.
The chlp is programmed by writing to the MAP of the
speech processor. To distinguish b~tween chip and MAP
writes, bits Aa - A12 ~re decoded. Any write to the block
1800 - 18FF in MAP will also write to the chip.
Two addres~es, one odd and one even (Y13 and Y14) are
decoded and ORed and the output of these (R/W new) can be
used to write to the filter chip in a more selective manner.
Odd writes are used to select the MU~ and even writes set the
auto sensitivity control (ASC) latch on that chip.
The four lowest address bitR are used ~o write to 14
r~gisters which contain the progra~miny information for the
acoumtic pracemsor chip. Reglsters YO to Y11 program
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W092/08330 2 ~ 9 3 3 4 4 PCT/AU91/00506
-2~- ..
irequency, Q, gain (attenuation or amplification) and
configuration in turn for each of the 3 filters. Register
Y12 sets the chip topology. Register Y15 is used to write to
the DAC.
Referring to Fig. 8, the topoloyy latch Y12 is as
follows:
D0 INV
. Dl, D2, D3 Topology bit~
D4, D5 DAC source
D6 DIR
DAC source
D5 D4
o 0 +5 volt
0 1 Audio
lS 1 0 Noise
- 1 1 ~xternal source
Topoloqv bits
D0 - INV inverts output of Filter 1
Dl - PARl sends Filter 1 output to summer
D2 - PAR2 sends Filter 2 output to summer
D3 - PAR3 sends Filter 3 output to summer
D6 - DIR sends the DAC output direct to summer
When a filter's PAR bit is set, the cascaded filter and
attenuator/amplifier are sent to the summer but the filter
output itself is sent to a bus so that it can be made
available to other filters. When the PAR bit is not ~et. the
filter-attenuator/amplifier combination is sent to the b~s.
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In this way the filter gains may be scaled if the filters are
. cascaded.
Filter ~rogramminq bits
Configuration latches Y3, Y7, Y11
D0, D1, D2, filter type select
D3, D4 clock select
DS, D6 f i I ter input
- Filter inPut selection
The filter inputs selected by D6 and D5 vary as shown
below
Input
Fllter O 1 2 3
1 DAC AUDIO FILT2 FILT3 <INVERTER
2 DAC AUDIO FILT1 FILT3
3 DAC NOIS~ FILT1 FILT2
; A clock pre-scaler is provided to extend the frequency
.. ranges of the filters. This is done by dividing the clock by
2 or 4 before feeding it to the filter's own divider.
Decoding is as follows:
. 20
D4 D3 DIVR Divider input
O O 1 5 MHz
O 1 2 2.5 MHz
1 0 4 1.25 MHz
1 1 1 5 MHz bu~ 2 switches (HF)
are opened in the filter to give double the centre frequency.
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W092/0X330 ~ a ~ 14 PCT/AU91/00506
-26-
Filter tvpe
.
With reference to Fig. 9, the filters consist of two
integrators in a loop with a variable gain feedback path.
The input may be a switched or an unswitched capacitor, it
may be applied to either the first or second input and the
output may be taken from either the first or second
integrator. This produces various different transfer
functions as given below.
Mode D2 D1 DO
0 SW~ 1st 1st Power down
1 SW 1st 2nd Lowpass
2 SW 2nd 1st Lowpass
3 SW 2nd 2nd Bandpass
4 US 1st 1st Highpac~
US 1st 2nd Bandpass
6 US 2nd 1st Bandpass
7 US 2nd 2nd Highpass
In this table a bit zero selects the particular
condition. Thus mode O, i.e. DO, D1, D2 all zero, powers
down the filter and switches ~ts output off.
In some cases it is desirable to shut all other
functions off. This is done by an external pin, PDB to power
.. down bimodal operation. Only the Ram Cell monitor is stiil
. operational.
The programmable filter clock divider comprises a 7 bit D
ripple counter which is compared wi~h the contents of the
frequency latch. The first match produces a counter reset.
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W092/08330 2 ~ ~ ~ 3 ~ 4 PCT/AU91/00506
--2
The frequency latch must be fed with the complement of the
count required. If for example a reset is required after a
count of 2 then the latch is all high except for bit D1 which
is low. The outputs of all the NOR gates connected to the
latch will be low except for the one connected to D1. Now if
the counter coun~s up from zero, on the first occasion of Q2
-~ going high, thls NOR will al50 go low and the 7 input NOR at
the output will reset the coun~er.
:: The filter Q is programmed by using a capacitor which
has a 4 bit binary sequence, toge~her wlth a 3 bit
programmable resistive divider. The resistors are programmed
by a number n which is represented by bits D4-D6 in the Y1
latch (YS and Y9 for fllters 2 and 3). The capacitors are
programmed by m, represe~ted by blts DO-D3.
The Q is glven by:
Q = 8( 1 ~ 1.R75n)~m
: 'By using switches, the ~eedback resis~or of an
amplifier can be con~igured to be like a buffered attenuator
- i.e. the output can drive another attenuator or lnverter.
Two separate sections are used, one to give ~, 4 dB steps
giving a total of +/- 28 dB and another to give 8, 0.5 dB
, steps with a maximum of +/- 3.5 dB. The total range is
therefore +/- 31.5 dB. Two 8 channel MUXes are used to
select the required taps on the potential dividers.
i: 25 The attenuator/ampli~iers can be selected by addressing
`.' latches Y2, Y6 and YlO. Bit D6 = 1 give attenuation, O
~' amplification.
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W092/a8330 PCT/AU91/005~
2~9~3~ 8-
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.o set the aain: Y2 = ain in dB ~; ~
To set the attenuation: Y2 = 64 - ~atten in dB ~ 2)
T~ Y2 is set to o (all bits low) then the
attenuator/amplifier is powered down and switche~ L-.
a Note that setting HF adds 6 dB to the gain.
The attenuator is arranged so that its gain is not
changed except when ~he signal passes through zero. This is
to prevent clicks and pops during gain changes. A zero
crossing detector which produces a pulse on either a positive
- or negative going zero crossing is used to strobe a latch
which transfers the required gain to the attenuator MUXes.
The multiplying DAC i5 a 6 bit resistive ladder type and
multiplies the input REF, selected by the SOURCE SELECT (Flg.
8) by the digital quantity in latch Y15. Blt D6 inverts the
output. The settllng time i~ 50 microseconds.
The acoustic aid processor 12, by its ~lexible and
programmable construction, allows many signal processing
strategies to be tried and ultimately settled upon so as to
best adapt the acoustic aid to provide information to the
wearer which compllments information received from an implant
aid worn in the other ear of the wearer. This same
flexibillty an~ programabillty alRo can be used to tailor the
bimodal processor for operation of an acoustic hearing aid
only.
In both cases it is the combination of use of a single
microphone together with the preprocessing capabilities of
~he speech processor 11 combined with the flexibility and
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-2g-
programability of the acoustic aid processor 12 which
provides features and advantages not found in hearing aid
devices to date.
The bimodal aid will now be described when used to
drive an acoustic hearing aid only. However the modes of
operation to be described in relation thereto arc equally
usable to help obtain the complementary behaviour me~tioned
above in the first embodiment in a relation to the use of
; both an acoustic aid and an implant aid by a singl~ wearer.
The following description therefore~~and to that extent should
be taken as applying equally to the first embodiment.
It should be under~tood that the nature of the
complementary behaviour between the two aids is subjective
and i5 determined by a combination of iterative testing and
wear~ng experience. The structure of the bimodal aid
described herein allows this complementarity to be achieved.
The testing procedures and methods for storing desired
patient parameters in the M~P memory storage 23 will be
described later in the specification.
2.0 SECOND_EMBODIMENT - BIMODAL AID USED AS ACOUSTIC AID
~` ONLY
The inherent flexibility of the acoustic aid signal
processor 12 incorporating the three software configurable
filters provides for an almost unlimited degree of
~lexibility of processing signals in the frequency domain
received from the speech processor 11 and destined for the
acous~ic aid 14. Four parti~ular modes of operation have
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W092/08330 ~ ~4 PCT/AU91/00506
~ 30-
been identified as desirable and achievable by the acoustic
aid signal processor 12.
The four basic modes of operation of the acoustic
output to the acoustic hearing aid are available and these
are shown schematically in Figure 11. ~ach mode encompasses
a large number of variations.
In mode 1, the filter parameters are set by the
audiologist during the iterative fitting procedure and remain
fixed thereafter. In modes 2-4, the speech parameter
~~ 10 extraction circuits provide instantaneous information about
the speech signal that is used to change the filter
parameters dynamically while the aid is in use. In modes 2
and 3 the input signal i5 the speech waveform. The output
signal is manipulated ~n dif~erent ways by controlling the
filter~ to emphasize the chosen parts of th2 waveform (such
: as the formants) and to attenuate other parts (such as the
background noise). In mode 4 the speech wave~orm is used
only by the speech parameter extractor, and the output
waveform is synthesized completely using the speech
parameters. The differences between thc original speech
waveform and the output of the hearing aid become greater as
one progresses ~rom mode 1 to 4, and the control over the
~requency spectrum and intensity of the output signal also
increases.
;25 2.1 Mode 1 - Frequency Response Tailoring
In this mode the acous~ic output is tailored to match
the patient's hearing loss. The 6 poles of filtering enable
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W092/08330 PCT/AU91/00506
-31-
.
this to be done accurately (usually within 2 dB of the ideal
gain specified by the audiologist at all frequencies) and the
Automatlc Gain Control allows the limited dynamic range of
the residual hearing to be used.
The acoustic signal processor of the second embodiment
configured in mode 1, provides both operational and
practical advantages over conventional hearing aids. These
advantages can best be appreciated by considering the steps
involved in setting up both types of hearing aid for
~~~ 10 operation: ~ ~ ~
; 'a) The conventional aid: The majority of
commercially available hearing aids merely amplify, and
sometlmes compress, the incoming sound. To f it one of
these aids the audiologist would normally measure the
user'~ threshold~ uslng an audiometer, calculate the
;j~ appropriate ideal gain by hand using a prescribed
fitting rule (e.g. the Nat~onal Acoustlcs Laboratory
. ~NAL) rule, Byrne and Dillon, 1986~. The audiologist
`~ would then search ~hrough the specifications of the
aids stocked in the clinic to find one with a gain ~hat
most closely resembled the ideal gain. On all aids
some changes can be made by the audiolo~ist although
the amount o~ control depends on the type of aid. The
feature~ that can be varied ~ay include any combination
o~ the overall gain, the maximum output, the level at
which compression begi~s. Frequency specific variation
of gain, if available at all, is usually only in two
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W092/08330 2 0 9 a 3 L~ 4 PCT/AU9l/005~
-32-
frequency bands corresponding to 'high' frequencies and
'low' frequencies respectively. Behind-the-ear aids
and body-worn aids, though less co~smetically
acceptable, usually of~er great~r scope for change by
the audiologist than in-the-ear aids. This is because,
with these types of aid, the acoustic properties of the
tube and earmould can be varied in addition to the
controls on the aid itself. In-the-ear aids also
require an earmould to be made speclfically for that
aid by the manufacturer. This is a costly and time
; consuminy business. This makes testing and comparing
in-the-ear aids dif~icult and expensive and many
clinics avoid using them. In-the-ear aid~ are also
usually more llmited in their maximum output and are
there~ore not aften ~uitable for more severe hearing
losses.
When ~he aid is configured it is then tested on the
client. If it proves unacceptable the audiologist must
,~ choose and recon~lgure another sort of aid. This is
repeated until an aid is found that the client
considers acceptable.
~b) The speech-processing hearing aid of the second
embodiment ~mode 1): the audlologist measures the
client'c hearing thresholds and any other hearing
levels ~hat might be needed for the strategies to be
tested, e.g. maximum comfortable level ~MCL). The
measurements are made usin~ ~he hearing aid and
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W092/08330 PCT/AU91/00506
-33-
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diagnostic and programming unit with associated
configuring software rather than a separate audiometer.
These values are then stored in a data file
automatically. A strategy is chosen and the aid is
configured accordingly taking a maximum time of about
five minutes. Calculation and fitting of ideal gain is
done automatically and can be quickly accessed in a
graphical form at any time by the audiologist. The
configured aid is then presented to the subjec~ for
- evaluation. Different f~ttings can-be tried in
quick ~uccession until an appropriate one is found.
' ~ence, the advantages are
; 1) The actual device, earmould and transducer are
used for mea~urement of client thresholds allowing more
accurate assessment of the ideal gain required for the
device. For conventional hearing aids these measures
are usually made using headphones and the effect of the
earmould acoustics is estimated separately. For in-
the-ear aids it is not possible to measure earmould
2Q acoustics before fitting because the mould and the aid
are manufactured together.
2) Different ~itting procedures (e.g. NAL
i ~ormulation, Byrne and Dillon, 1986) can be implemented
for testing very quickly without requiring a change of
~5 a~d because the changes are programmed in software
rather than by hardware adjustment.
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W092/08330 2 ~ 9 3 3 ~ 4 -34- PCT/AU91/00506
3~ The gain fi~ting can often be more accurate
than is possible on many commercially available aids
because of the flexible programming of frequency
responses. These values are then stored in a data file
automatically. A strategy is chosen and the aid is
configured accordingly, taking a maximum time of about
five minutes. Calculation and fitting of ideal gain is
done automatically and can be quic~ly accessed in a
graphical form at any time by the audiologist. The
configured a1d is then presented to the subject for
evaluation. Different fittings can be tried in quick
succession modelling various available aids because of
the flexible programming of frequency responses.
4) The audiologist can change the ideai gain
~unction at will if he/she believes that the ideal gain
based on the client's threshold measurements is not
optimal for that client. With many conventional
hearing aids th1s can only be done gro sly by changing
the gain in "high" or "low" frequency bands or by
choosing a different aid with quoted specifications
closer to the new requirement for the client.
5) Information about the fitting is available to
the audiologist at any stage thus giving them more ~'on-
line" control over the fitting than with any aid on the
mar~et.
6) The calculation of ideal gain is done
automatically for most fitting procedures, (with the
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W092/08330 2 ~ 9 ~ ~ ~ 4 PCT/AU91/00506
-35-
exception of those used on insertion gain bridges, this
has to be done by hand) and thus the new device saves
time and removes a possible source of error.
In summary, the device of thP second embodiment can be
configured exactly as many conventional a:ids, often more
accurately. Setting up and testing the device are quicker,
more efficient, and less prone to sources of error.
Fig. 12 provides an example f~tting in mode l which is
achievable utilising the acoustic aid signal processor 12 of
. lQ the bimodal speech proce~sor. - - ~
2.2 Mode 2 - Loudnes~ MaPpinq
This is similar to mode l except that the level output
at any ~pecl~ic frequency ~s mapped non-linearly on a
~requency speclfic baqis by dynamlcally changing the gain
parameters of the three fllters in response to amplitude and
; frequency variations measured by the processor. This
requires that the audiologist measure the maximum comfort
levels to which maximum amplltude can be mapped in addition
to client thresholds. The advantage of this mode is that it
makes a more accurate mapping of dynamic range possible.
Hence, if used appropriately, the relative loudness of the
spectral components i9 preserved. This may be better than
mode l for users whose dynamic range changes a lot as a
function of frequency. This method of loudness control avoids
many of the undesirable spectral distortlons that accompany
more commonly used schemes such as peak limiting and non-
11near corpr~ssion.
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W092/08330 ~ ~ PCT/AU91/00506
-36-
2.3 Mode 3 - Dynamic Enhancement of S~ectral Features
In this mode the frequency paramet:ers of the three
filters are changed dynamically (unli~e modes 1 and 2 where
they are fixed). When the values are made to chanse in a
manner depending on the speech parametets measured by the
processor then salient speech features can be enhanced. This
gives rise to a wide range o~ possible speech-processing
strategies in this mode. For example the centre frequencies
of two bandpass filters can be used to track the F1 and F2
(first and second formant) peaks~~in-the signal. This acts as
both a form of noise cancellation and also a removal of parts
of the signal that migh~ mask the information in the peaks to
be traced. The resulting signal a~ter flltering :Ls amplified
to the appropriate loudness for the user on a frequency-
specific basis as in mode 2. Thlq may be most u~e~ul ~orusers with impaired frequency resolution as well as raised
thresholds. The device used in thls mode can also be used to
amplitude modulate the signal at the fundamental frequency
(FO) which can be another way of enhancing this parameter.
The most similar commercially available device~ are the
"noise-cancelling" hearing aids such as those containing the
"Zeta-noise-blocker" chip. These devices calculate an
average long-term spectrum that represents the background
noise and this noise is then filtered out of the signal along
with any speech that happens to be at the same frequencies.
This mode 3 scheme i~ based on enhancement of the speech
.
signal at the measured formant frequencies rather than
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W092/08330 ~ 9 ~ 3 ~ 4 PCT/AU91/00506
-37-
cancellation of noise. This means tha~ speech information
which is close in frequency to the noise will not be lost
although the noise further from the formants will be reduced.
The schemP will also enhance the seiected speech features in
quiet conditions as well as in noisy conditions.
Fig. 13 provides an example of mode 3 wherein selective
peak sharpening is performed by the acoustic aid signal
processor 12.
2.4 Mode 4 - SPeech Reconstruction
- This mode differs from the-other-modes of operation of
the second embodiment in that the user does not receive a
modified version of the input signal, but a completely
synthesized signal constructed using parameters extracted by
the speech processor. The signal can be reconstructed in
many different ways depending on the u~er's hearing loss.
This reconstruction provides very tight control over the
signals presented and hence allows very accurate mapping onto
the user's residual hearing abilities. It may be most useful
for users with very limlted hearing for whom normal
amplification would provide no open set speech recognition.
A second example of the use of this mode is for
frequency transposition. Sounds normally occurring at
frequencies that are inaudible ~o the user can be represehted
by synthesized signals within the audible range ~or that
user. Such schemes have been attempted ln the past, but not
using a completely re-synthesized waveform as in the present
case. The re-synthesis scheme has been shown to work for
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W092/08330 2 Q ' ~ '~ PCT/AU91/OOS06
-3~- ~
electrical stimulation with cochlear implant users and may be
of benefit to severely-to-profoundly impaired hearing aid
users as well.
Each mode of operation allows a wide range of potential
strategies. The modes are not discrete ,and some strategies
that combine elements from different modes can be
implemented. For example, a reconstructed signal
representing FO information (mode 4) can be added to a
filtered speech signal ~mode 3).
; - 10 3.0 USE OF THE BIMODAL AID -
Wlth reference to Fig. 2 the bimodal aid is proyrammed
by use of a diagnostic and programming un$t 21 which
communicates with the speech proceqsor 11 and, in turn, with
the acoustic aid processor 12 by way of a diagnostic
programming interface 22. The diagnostio and programming
unit 21 is implemented as a program on a personal computer.
The interface 22 is a communications card connected on the PC
bus.
Software has been written to find the optimum filter
settings to produce the frequency~gain characteristic
specified by the audiologist, for use in the frequency
response tailoring mo~e of operation described above.
So~tware to pro~ram to the other modes of operation has also
been pr~grammed and ~ested.
With ref*rence to Figs. 14, 15 and 16 the basic
procedure for use of the bimodal device is a3 follows.
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In bimodal use the fitting procedure is as outlined in
; flow chart form in Fig. 14. The bimodal MAP is produced on
the personal computer following an iterative testing
procedure of the subjective performance of the bimodal aid
S for a multiplicity of trial settinys of the MAP.
Fig. 15 outlines the procedure in flow chart form withparticular reference to obtaining an op~timum setting for the
. acoustic ald 14 in mode 1.
Fig. 16 outline~ the basic interaction between the
10- control program in the diagnostic and programming unit and
the fitting and mapping procedures performed by the
. Audiologist.
The above de~cribes only some embodiments of the
present invention and modifications obvious to those ~killed
in the art can be made without departing from the ~cope and
splrlt of the present lnvent1on.
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