Note: Descriptions are shown in the official language in which they were submitted.
29
Il
., . .
,~ BACKGROUND OF THE INVENTION
l. Field of th~ Invention. This invention is directed to systems and
, devices ~Yhich are useful in the improvement of hearing ability, in general,
, and, more particularly, to methods and apparatus for providing improvement in
he2ring and spatial processing of sound by improving the discernment of sound
~i in the "perceptual scacel' of the individual.
2. Prior Art. It is currently recognized by the public at large that
¦ hearing impairment is a serious problem. However, this problem has,
j generally, not received the same attention as other diseases, maladies or
i impairments. Typically, the reason for the lack QT attention is that hearing
impairment is ~he "silent handicap". That is, it is not as readily apparent
I to the public as are other physical handicaps. In fact, many hearing impaired !
,¦ individuals are unaware of their loss until tested or confronted with a
i specialized circumstance. Nëvertheless, impaired hearing can haYe a
il significant impact on the quality of life of the individual involved.
! Therefore, it has been a source of investigation by many researchers (of
I various levels of ability) over the years to produce hearing enhancements or
"hearing aids". These "aids" are available at various levels of technical
l expertise.
, One type of hearing aid available on the market uses noise supression
techniques. However, conventional filtering techniques generally are not
considered to be effective or adequate for providing truly high fidelity
frequency compensation which is desirable in hearing aids. Thus, results from '
Ij implementation of these techniques often suffer from muffled sound outputs,
!¦ and unac~eptable noise and ringing problems.
'I A further problem in the conventional design of hearing aids is the
inadequate treatment of background noise. Thus, a related problem with
conventional hearing aid design is that the user will normally reduce the
' volume to reduce the higher intensity energy produced, for example, by
I vowels. However, at the same time the user sacrifices speech intelligibility
by simultaneously reducing the intensity of the lower energy signals, e.g. I. ~ ,.
4~29
!I sounds produced by consonants- Further, hearing aids which employ automatic
gain control (i.e. decrease gain as input level increases) have the
¦¦ disadvantage of decreasing the gain as a function not only of the lower
Il frequency, stronger vowel sounds contained in speech but also by the large
l¦ energy, low frequency background noises. Because background noise and vo~els
can have the same effect on the gain control, an abnormal relationship between
speech sounds is introduced. High frequency consonants~ for example, are not
amplified sufficiently in the presence of background noises thereby resulting
l in greatly reduced speech intelligence. In conventional hearing aid systems
¦ all sounds are amplified whereupon background noises greatly mask speech
~¦ intel~ bility.
¦~ It is well known from Bekesy's model of the ear that predominantly low
frequency noise masks the higher frequency consonants because of the
~ travelling wave phenomenon of the basilar membrane, i.e., low frequency
information masks high frequency information, whereas, the reverse is not
true. This phenomenon is commonly referred to in the literature as the
"upward spread" of masking.
A particularly troublesome area for the hearing impaired occurs during
normal conversation in an environment of a conference or large office.
Persons with normal hearing are able to selectively listen to conversations
from just one other person. The hearing impaired person has no such abilit~
and, thus, the individual experiences a phenomenon known as a "cocktail party
effect' in which all sounds are woven into an undecipherable fabric of noise
and distortion. This condition is aggravated for the hearing impaired because
! all incoming sounds have a single point source at the output transducer of theconventional hearing aid. Under these circumstances, speech itself competes
with noise and the hearing impaired person is constantly burdened with the
mental strain of trying to filter out the sound he or she wishes to hear. The
I result is poor communication, frustration and fatigue.
3~ , Yet another performance shortcoming of the conventional hearing aid,
I particularly in "open mold" hearing aid fittings, resides in the area of audio
.,
L~ 9
feedback. The ampliEied signal is literally routed bacls to
the hearing aid input micropllone and passes th~ough the
amplification system repeatealy so as to produce an extremely
irritating whistling or ringing. While feedback may be
controlled in most ~ixed listening situations, it has not been
controllable for the hearing aid user who faces a changing
acoustic environment.
Another area of hearing impalrment, related to background
noise, is experienced in many noisy environments. These
environments include industrial locations, office areas,
computer rooms, airport pad locations, to name just a few.
In these environments, even persons with "normal" hearing
experience difficulty in understanding and/or discerning
sounds, whether vocal or otherwise. That is, normal convers-
ation is impossible and persons must shout to each other merely
to be heard. Moreover, in many of these environments
(especially industrial or airport locations), persons wear ear
protectors to prevent damage to the ears. In fact, in some
instances, such ear protection devices are mandated by law.
In these cases, a standard hearing aid is of little or no
advantageous consequence, for the reasons discussed above.
However, it is highly desirable to have some type of hearing
enhancement device or apparatus for use in these situations
for comfort, convenience and/or safety.
CROSS-REFERENCE
Reference is hereby made to U.S. patent 4,658,426 issued
April 14, 1987 to Douglas M. Chabries.
PRIOR ART PATENTS
Reference is made to the following U.S. Patents which are
listed in Patent No. order.
U.S. Patent No. 4,238,746; ADAPTIVE LINE ENHANCER; McCool et al.
U.S. Patent No. 4,349,889; NON-RECURSIVE FILTER HAVING
ADJUSTABLE STEP-SIZE FOR EACH ITERATION van den Elzen et al.
4~
.. '11 . I
¦ U.S. Patent No. 4,243,935; ADAPTIVE DETECTOR: ~cCool et a].
j U.S. Patent No. 4,052,559; NOISE FILTERING DEVICE: Paul et al.
U.S. Patent No. 4,038,536; ADAPTIVE RECURSIVE LEAST MEA`~ SQUARE ERROR
l FILTER; Ferntuch.
I U.S. Patent No. 3,375,451; ADAPTIVE TRACKItlG NOTCH FILT R SYSTE~I; Borelli
et al.
U.S. Patent No. 4,302,738; NOISE REJECTION CIRCUITRY FO~ A FREQUENCY
DISCRIMINATOR; Cabot et al.
l U.S. Patent No. 4,480,236; CHANNELIZED SERIAL ~DAPTIVE FILTER PROCESSOR;
~ Harris.
I SUMMARY OF THE INVE,ITION
I .... _._ . .. _. .
This invention is directed to a method and apparatus for improving the
hearing capability of persons with some type of impaired hearing, whether
implicit or imposed. The invention comprises a system which empirically
detects the portions of a person's hearing ~hich are impaired. The hearing
aid system is then particularly selected to enhance those impaired portions.
This may include a reduction in some impairments which are in its nature of
over sensitive hearing capability. The entire process and apparatus of this
invention is directed at enhancing the overall hearing capability of the
person in that~person's "perceptual space", thereby to produce an improv;ed
hearing signal at the auditory nerve. The invention d~s not merely amplify
all sounds.
Ihe invention provides for noise suppression, feedback suppression,
frequency compensation and recruitment. These improvements can be supplied
together or separately and in any order. By using all of these improvements~ I
the optimum signal can be obtained. However, a lesser signal can be produced
by using less than all of the improvement techniques.
The invention uses a transmultiplexer which, essentially, separates the
incoming signal into a plurality of bands. These bands 2re then operated upon
t separately. Appropriate suppression is achieved by adaptive filters,
multiplication circuits or the 1ike. Other operations such as taking the log
.
xs
and the exponential of the signals are used to "map" the prescribed
apparatus for the individual aid. The several bands are then recombined
to produce the output signal which is supplied to the individual.
In the context of this description, the phrase "hearing aid" or
"Hearing enhancement device" is intended to include an apparatus or
device which is used to enhance the hearing capabilities of a person
within his (or her) environment. It includes but is not limited merely
to devices for assisting those persons with individual hearing
impairments.
BRIEF DESCRIPTION OF T~E DRAWINGS
Figure 1 is a graphic representation of an auditory area for a
person with "average" hearing.
Figure lA is another graphic representation of the dynamic range of
"normal hearing" persons as measured in response to pulsed narrow bands
of sound.
Figure 2 is a graphic respresentation of the relationship between
loudness in sones and loudness level in phons of a 1 KHz tone.
Figure 3 is a block diagram of a model of a typical hearing
operation.
Figure 4 is a block diagram of a model of the hearing enhancement
device of the instant invention.
Figure 5 is a block diagram of a transmultiplexer apparatus of the
instant invention.
Figure 6 is a block diagram of a noise suppression device with a
delay in the transform domain, which can be used with the instant
inventlon .
Figure 6A is a schematic representation of a three-tap FIR filter
which can be used with the instant invention.
Figure 7 is a block diagram of a noise suppression device with a
delay in the time domain.
Figure 8 is a block diagram of a noise suppression device using a
constant primary input value.
Figure 9 is a block diagram of a feedback suppression device which
can be used with the instant invention.
Figure 10 is a schematic representation of one embodiment of a
iZ~3
frequency compensation network which can be used with the instant
invention.
Figure 11 is a graphic representation of a recruitment
characteristic as related to a "look-up table" which can be used with the
instant invention.
DESCRIPTION OF A PREFERRED EMBODIMENT
Referring now to Figure 1, there is shown a typical graphical
representation of a "normal" hearing pattern for the "average" human ear.
In particular, contours of equal loudness (phons) are plotted against the
intensity level (in decibels) and frequency in Hz. In this instance, the
contours are numbered by the equal loudness correspondence with the
intensity level at 1000 Hz. It should be noted that the contours of
equal loudness are, typically, spaced logarithmically and, hence,
annotated in decibels (10 log10). The human hearing system must account
for this non-linearity.
In this graph, contour 0 is defined as the threshold of hearing.
That is, below this intensity the normal human ear does not perceive
sound. Thus, at O dB and lO00 Hz, a sound is just barely audible to the
average person. On the other hand, at 50 dB and 1000 Hz the sound is
well within the normal hearing range. Conversely, even at 40 dB, a 50 Hz
signal normally is inaudible.
At the other end of the range, the upper contour is referred to as
the threshold of pain discomfort. That is, the application of a signal
of appropriate freqency at or above the designated decibel level will
produce discomfort (pain) and, perhaps, damage to the ear. It is seen
that this threshold of discomfort pain remains fairly constant at a level
of approximately 125 dB.
However for hearing aid fitting, a "loudness discomfort level" (LDL)
should be employed as an upper limit for hearing aid output rather than a
threshold of pain. By following this approach, it is possible to avoid
actual pain, discomfort (in the hearer) due to loudness, the introduction
of non-linear distortion by overdriving the basilar membrane, and/or
physical damage to the parts of the inner ear.
Figure lA shows a graphic presentation of the sound pressure level
(SPL)
ll v c~ :`
~¦ N2 frequency. This Figur~ also shows the mean and the range for comfortable
¦ (MCL) and uncomfortable listening levels (UCL) for pulsed narrow band noise.
¦ Subtracting the thrèshold le~els from the upper range for tne UCL, provides
I the dynamic range of hearing for "normal" hearing persons. Thus, bet"een 25Q
, and 8000 Hz the dynamic range is between about 80 and 95 dB.
¦I However, it has been determined that in many instance~ or' hearing
impairment, this dynamic range is significantly altered. Impairment of
hearing occurs when the threshold of hearing for an individual is,
¦ effectively, raised. Thus, the dynamic range for that individual is reduced
and possibly distorted. Moreover, it may be that the threshold of hearing is
increased uniformly as a function of frequency. If the threshold of hearing
is, in fact, increased uniformly across frequency, the typical approach to
hearing aid construction, i.e., the mere amplitication of the signals, will be
beneficial. However, it is clear that even with a uniform increase o,
threshold of hearing, a uniform amplification thereof will amplify both
desired frequencies (where a hearin3 loss exists) and undesired frequencies
(where hearing is normal). This operation is, oF course, recognized as a
critical problem with conventional hearing aids currently available.
However, it is recognized that the hearing impairment that is most
typically encountered is not merely a uniform rise in the threshold of
hearing. More typically, what occurs is an alteratlon in the shape nf the
¦ threshold hearing contour wherein certain frequency ranges are not received as ¦
well, or at all.
It is the purpose of this invention to recognize that the human hearing
system can be modeled as a non-linear process ~lith measureable dynamic range
and pass bands and, further, to provide a hearing aid which is programmable
¦ and which exploits this non-linear hearing model to compensate for each user's¦ particular hearing loss in such a way as to reduce distortion, improve the
i signal-to-noise ratio, yield improved speech intelligibilit~ in the presence
of noise including speech babble, reduce or eliminate audio feedback and
¦~ provide output between the threshold-of-hearing and the
.
,q~ ~
threshold-of-discomfOrt (LDL) contours for all requencies. Similarly, the
¦ invention enhances loudness perception to the hearer.
¦ The relationship of loudness in sones to loudness in phGns for the normai
I, ear is shown as the solid line 2A in Figure 2. This is a l~g/lo~ plot where
i 40 phons equals 1 sone. Recruitment, an abnormally rapid growth in loudness,
i is represented by the dot-dashed line 2B for an individual ~;ith a 50 dB
hearing loss at 1 KHz. That is, this individual cannot hear be1cw 50 dB.
¦ However, the loudness grows rap;dly until at 6~ dB and i sones the loudness
, perception of the person is equal to that of a normal hearin~ system. This
I non-linearity must be taken into account for the he2ring impaired listener.
The type of hearing impairment which is encountered ~y ciffer~nt
individuals varies~ The conventional hearing àid which is currently available
¦ on the market is simply not adequate for all persons. I
¦ Referring now to Figure 3? there is shown a functiona1 block diagram which !
lis representative of a non-linear model of the hearing operation of the human
I hearing system 300. In this arrangement, sound is provided by a typical
source 308 and received in the ear apparatus. The ear oper2tes as a frequency
transducer 301 which separates the incoming sound signal into a plurality of
band pass output signals A. These band pass output signals are supplied to a
transfer function 302 which operates to enhance the band pass output signals
`~ by increasing or decreasing the amplitudes of these signals In this way, the
\ ear can selective1y reject background, or noise, sinals and concentrate on
the desired signals.
The signals Bfrom the transfer function 302 are proYidec to the log
I circuit 303 which performs a logarithmic function therecn. The oucput C of
the log function 303 is supplied to the recruitment func~ion 304 which,
¦effectively, scales the supplied signals as a function of frequency to produce
¦ an output with a dynamic range which fits between the thres~old-of-hearing andI the threshold-of-discomfort (i.e., the dynamic range of the ear) for all
!hearing range frequencies.
i The output D of the recruitment function 304 is supplie~ to the clipping
!
. , ,
(~ ~
~1 '
¦ or saturation function 305 which has the effect of cutting off extremely low
¦¦ and high amplitudes by saturat;ng. The output E of the clipping function 30S
¦¦ is provided in what is referred to as the "perceptual space" 305. This
~ perceptual space is, for purposes of this discussion, defined as the signal
S !i space at the input ends of the auditory nerve. The effect that is produced by
the hearing system is, essentially, the mappin3 of signal~ to the auditory
nerve input, which will then simulate nerve firings or the like, which can
then be detec~ed as appropriate sounds.
l For this invention, then, it is understood that the hearing operation an~
¦ the impairment thereof is a function of the operation or one or more of the
¦ functions shown and described in the "dual" of the human hea~ n~ system shown
in Figure 3. For example, if the sensitivity ,~unction 302, the log funct-ion
303, the recruitment function 304, or the clipping function 305 is, in some
way defective, a portion of the band pass sign~ls supplied by the frequency
transformation function 30l are lost, diminished, enhanced, or the like. Thi
loss can be produced at signal level A, B, C, ~ or E. Any such deformation of
the hearing function will, of course, produce an undesirable impairment of the
hearing as detected at the perceptual space 306.
While the dual described above in relaton to Figure 3 is believed to be
accurate, it is to be understood that modificarions to this dual can be made
by combinin~ functions, separating functions, re-derining or f ne-tuning
functions, and so forth.
As shown in Figure 3, a hearing enhancement device lO0 can be interposed
between the sound source 308 and the mechanism 300 ~hich represents the human
hearing system. This hearing enhancement device lO0 is shown in dashed
outline, to indicate that it is separate from the actual ear mechanism, and
that it is supplied only in those instances where necessary.
It is presumed that when the hearing system 300 operates in the normal
¦ fashion (as suggested relative to Figures l, lA and 2), a hearing enhancement
device lO0 is not neressary. In the event that the hearing system 300 is not
functioning properly, the hearing enhancement device lO0 is insertèd into the
hearing processing channel.
,~
LlL~
In the present invention, the hearing aid device 100 is used in an
attempt to compensate for any deficiencies in the actual hearing
mechanism 300.~ In a typical application, the individual is tested, in a
empirical fashion, by applying sounds at various frequencies to the
individual by means of an audiometer or the like. The results of these
tests can produce a transfer characteristic for the ear as shown in
Figure 2, together with the information for the auditory dynamic range as
shown in Figure 1 and lA. By utilizing these characteristics, the
hearing aid device can then be programmed for the individual in a
prescription-like basis.
Referring now to Figure 4, there is shown a schematic representation
of a system incorporating the hearing aid of this invention. In this
Figure, there is shown an apparatus which receives sound wave signals at
the input of band pass filter 401. The filter is arranged to produce a
plurality of band pass frequencies which are separate and substantially
independent. That is, there is little or no overlap of the frequencies
in the respective "bins" which are defined by the band pass frequencies.
Typically, these filters can be symmetric band pass filters evenly spaced
across the bandwidth of the input signal. Likewise, in an efficient
implementation the number of filters is an integer power of two. Also,
it is assumed that the number of filters (and their shapes) provide
sufficient frequency resolution such that any desired transfer function
can be realized as a weighted sum of the filters.
These multiple band pass signals are then supplied to the processing
25 circuit 403, the logarithmic circuit 404, the recruitment circuit 405 and
the saturation circuit 406. These circuits or devices operate in the
same fashion as those devices which were desribed relative to Figure 3.
However, it is noted that the human hearing system 300, i.e. the
operational capability of the individual, has previously been tested in
accordance with the system shown in Figure 3. As a consequence, the
shortcomings or impairments in the hearing process have been detected and
appropriate compensation can now be made. This compensation can be made
by inserting nverting networks into the hearing aid system. Thus, an
inverse recruitment stage 407 is used to provide compensation for the
35 recruitment stage 405. The output of the recruitment stage 407 is
supplied to the exponentiating circuit 408 which has the effect of
compensating or negating the log circuit 404.
In similar function, the sensitivity circuit 409 is the inverse
sensitivity circuit 403 and compensates for the operation of processing
circuit 403.
The output of the system includes a reconstruction device 410, which
is, of course, the inverse of the banded band pass filter 401 noted
above. The reconstruction device 410 re-combines all of the band pass
filter signals and supplies the ultimate combined sound signal. This
output is used as the hearing enhancement device 100.
Additionally, digital signal processing techniques for ~eedback
suppression and/or noise suppression are also applied to the signal.
Application of these techniques is most effective at the output of the
recruitment circuit 405 or the saturation circuit 406, but may be used at
15 the output of processing circuit 403 or log circuit 404. Previous
techniques for noise suppression have applied these algorithms to the
unprocessed acoustic signal and have provided an output with a muffling
effect, thereby reducing the intelligibility of speech signals. Recent
noise suppression algorithms have attempted to correct for this muffling
effect. Specific embodiments of the noise suppression and feedback
suppression are described as part of the invention. A further property
of the processing described is that linear phase may be retained to allow
binaural processing.
It has been determined that the precise order of the processing
circuits between the input filter 401 and the reconstruction or output
filter 410 can be varied. Moreover, one or more of these processing
operations can be omitted if desired or required for some purpose.
However, by removing one or more of the processing circuits, the signal
processing ability of the system is reduced, whereupon the output signal
supplied is also reduced in content.
Referring now to Figure 5, there is shown a block diagram of a
transmultiplexer system 500 which performs in accordance with the instant
' invention. As shown in Figure 5, the transmultiple~er is,
Il essentially,comprised of the five component p~rtions including the input
¦I pre-filtering stage 501, the time-to-frequency transforms ~FFT) 502, the
li processing blocks in the transform space 503, the frequ2ncy-to-time transforms
, (inverse FFT) 504, and the output post-filterin3 st2ge 505. The processing
¦I blocks include a noise supression stage 50~, a ~eedback supression stage 507,iI a frequency compensation stage 508 and a rëcruitment stage 509.
The transmultiplexer 500 operates on the basis of an al~orithm which
I transforms a time signal to its frequency represent~tion at stages 501 and
l¦ 502, allows independent processing between ,requenc~ bins in the transform
¦ space 503, and then transforms the frequency re?resent2tion back into a time
Il signal (stages 504 and 505) In the digital hearing aid, the transmultiplexer
li is used to maximize the homomorphic processing ?otential in the transform
I space 503 by assuring that the bins in the transfor~ space are essentially
I independent.
In general, an FFT is a computationally efficient algorithm for obtaining
the frequency representation of a time signal. The output of an N point FFT
is N frequency bins, each appro~imating the amplitude of the time signal in
j that frequency ranger ~owever, the value in a par-icular frequency bin is not
a function of the energy at that frequency alone, but, rather, there is a-
~l significant interaction between the actual energies in seve al adjacent bins.¦! Inasmuch as the values in the bins are not inde?endent one bin cannot be
scaled without affecting other frequency bins ~nen the inverse FFT function is
l performed. In a preferred embodiment, the transmultiplexer algorithm uses two
¦ overlapped FFT's, as well as input and output filteri"g, to decrease
¦I dependence between frequency bins. The frequenc~ bins do not overlap
¦¦ significantly with bins adjacent thereto.
As stated, two overlapped FFT's are required in this im~lementation of the
¦ transmultiplexer. In this embodiment, the inpu,s to each FFT 502A and 502B
1l are the outputs of t~Jo separate input filter banks 5QlA and 501B,
Il respectively. The input filter banks have the same coe,ficients but the input
~LV~ 9
signal supplied to one of the banks (e.g. bank 501B) is passed through delay
network 510 and, thus, delayed by half the number of filters in the banks. In
particular, where N is the number of filters in the banks, the input to bank
501B is delayed by N/2 samples.
The output filters are the same as the input filters except that the
filter coefficients are arranged in a different order. These coefficients are
provided by a different sampling of the ~lindow function noted above. Also,
the output signal from filter bank 505A is passed through delay 511 and
Il delayed by N/2 and then added to the output signal of filter bank 505B at
IO 1¦ summing junction 512 to yield the processed transmultiplexer output. Thus,
il the system accomplishes an over,ar) and-add structure. The inputs to the two
output filter banks 5~5A and 505~ are the outputs of the t~lo overlapped
inverse FFTs 504A and 504B. The alsorithms of FFT 502 and inverse FFT 505 are
well documented in the literature and need not be discussed here. It should
be noted that, in a preferred embodiment, the actual computations required in
the transforms, as well as the computations in the intermediate processing
blocks, can be cut in half by taking advantage of the symmetry of the FFT.
As shown in Figure 5, a variety of functions can be performed on the
signals in the transform space 503. These operations include noise
suppression, feedback suppression, frequency compensation or equalization, and
i! recruitment~ Inasmuch as each oF these operations can be performe~ as 2
¦I separate function, different combinations and arrangements thereof can be used
in order tp correct for specific hearing disorders in the context of the human
l hearing system model 300. Figure 5 presents an optimum system in which all of
¦¦ the above mentioned operations are included.
! There are many ways to implement noise suppression, in particular a
tl frequency domain adaptive noise suppressor. One implementation of a noise
suppressor 506 is shown in Figure 6. The noise suppressor comprises a bank af
il adaptive filters 601. Each of the adaptive filters includes a FIR filter 602
3~ ¦I with feedback 603. There is one filter per bin thereby realizing the symmetry
savings noted above. Each filter 601 may include a different ~1 forming a
vectcr ~ when considering all filters in the filter bank. The ~er+a!~ r~
¦I permits control of the adaptation times in the fre~uency bins. ~f noise
suppression is employed at the input to band p25S filter 401 or processlng
circuit 403, in the system of Figure 4, then the ~ for each frequency region
wi11 be different to ai'r~w equal adaptation times. If noise suppression is
¦ applied at the output of the functions 403, 405 or 406, then a single ~ can
suffice. These adaptation times can be experim~nta,ly dete,mined and an
optimized ~ can be found for each embodiment. The bul~ delay 603
incorporates a delay time z A and is used to decorrelate the "primary" input
¦ to the filter with the "desired" response. The delay time, f~ , in this
¦¦ embodiment is equivalent to ~ x N/2 samples. This permits noise suppression
in the adaptive filter.
! Referring now to Figure 6A there is shown a schematic representation of
one of the filters used in the input and output filters banks of the 3-weight
¦ Finite Impulse Response (FIR) filters 505 shown in Figure 5. The output of
one of the filters is given by the simple equation:
Output (j)= (a ~ b z 1 + cZ 2) Input (j)
= a.Input(j) + b~Input (j-l) +c-Input (j-2)
where a, b, and c are constant filter coefficients, the subscript j indicates
sample j and z 1 jS the standard notation for a unit sample delay. These
coefficients ar~e selected a$ noted above.
The coeffi¢ients for the filters are samples frcm a ~lindow function which
modifies the input signal so the bins in the szmple space will not overlap.
I Any ~Yindow function can be used so long as the function insures that the bins¦ are not aliased. The decimation of the input signal depends on the number of
FIR filters in the filter bank. For ev~ample, in a filter bank with 16 ',
filters, every 16th sample would be gated to a particular filter, i.e. filter
1 receives samples 1, 17 and 33, and so forth.
Alternatively, as shown in Figure 7, the noise supressor 506 can also be
implemented by inserting the delay 703 between the inputs of the filter
¦ banks~ Mathematically, this puts the delay in the time domain, dnd requires
I transforming this delayed signal into the trans,orm domain. The delayed input
., !
I, .,
L~
signal is transformed in the same manner as the undelayed signal with two
overlapped FFT's 701, 702 preceeded by two FIR filters 704, 705. The
input to the delayed signal filter bank 706, 707 is delayed by ~
samples from the main input. The output of the delay FFT's 708, 709 is
then used as the primary input to the noise suppressor 725
which is a representative circuit arrangement. The output is Y.
With this method of noise suppression, each frequency bin is
multiplied by some attenuation factor Ak(m). This attenuation factor is
determined from the smoothed powder (i.e. the average power in the bin)
and the estimated noise power in each bin. The attenuation factor is
determined by the frequency bin, the sample number, the estimated noise
power, the smoothed power, and the square of the magnitude of the
amplitude in the selected frequency bin and the circuit follows the
equations:
Ak(:l)=l-[Nk(l~l)/Xk2(~)] and
~Ik (~1)= Xk(M~-Pk(l~l)
where k denotes the frequency bin, M denotes the sample numb~r, N2 is
the estimated noise power, x2 is the smoothed power, and Pk is the
square of the magnitude of the amplitude in frequency bin k.
The implementation shown in Figure 7 requires six FFT's per block
(N samples) as compared to four FFT's per block when the bulk delay ~ in
Figure 7 is transformed into the transform space 503. In this
embodiment, the delay time is equivalent to ~ samples in the time domain.
This will create a real-time performance requirement due to an increase
in computation as compared to the system using four FFT's.
Another method of noise suppression is shown in Figure 8. This
embodiment assumes a constant noise value in each of the frequency bins.
Typically, this value is set to 1. The constant value C is the primary
input to the adaptive filter 800. This type of noise suppression is also
called spectral subtraction.
The methods of noise suppression described herein use the same basic
adaptive filter which is well known in the art (as are the output and
update equations thereof).
16
L~2 ~J
Referring now to Figure 9, there is shown a schematic diagram for
one embodiment up the feedback suppression stage 507. The feedback
suppression function is produced by a feedback suppressor comprised of an
adaptive filter 901 governed by the same equations as the noise
suppressor. However, the bulk feedback delay 902 for the feedback
suppressor 507 is greater than the delay for the noise suppressor and is
chosen to decorrelate speech. Typically, the delay is about 100
milliseconds. Also, the output of the feedback suppressor is defined by
the Error signal.
Figure 10 is a schematic representation of one frequency
compensation network. The frequency compensation stage 508 corrects
the frequency spectrum of the input signal from the band pass filters
401, for example. The exact correction required for the frequency
spectrum is determined for each individual. Typically, this function
will be measured by audiologists. In its simplest form, the equalization
is performed by multiplying the output of each frequency bin by some
scale factor K which is the frequency correction scaler for specified
transform bin. The various scale factors K will be selected for each
individual thereby assuring a good "prescription" fit.
Recruitment is the phenomenon which accounts for the non-linearity
of an individual's perception to a linear change in sound amplitude.
Recruitment is a means by which the transform bin power is mapped
into a region bounded by the threshold of hearing and the threshold-
of-discomfort. This mapping of the bins is inherently non-linear and
may be accomplished in several ways. One appropriate approach is
through a "takle lookup", with one table for each bin. The table
contents are scale factors, much like the frequency equali~ation scale
fators, and are determined by individual testing. Figure 11 is a
graphic representation of a typical recruitment characteristic 1100
for an individual. This sample curve is not intended to represent
any specific characteristic. However, the several points on the
curve are representative of the information which will be stored in
the look-up table. Thus, when a particular "input" is received, the
recruitment device 509, for example, will produce the appropriate
"output". This output will be appropriate to enhance
- - (
!l
the individual's hearing within the prescribed dynamic range. Thus, the
actual hearing capability of the user is enhanced and optimized.
Thus there is shown and described a new and unique approach to the concept
I¦ of hearing enhancement. By the approach physically impaired hearing can be
¦ im~,oved. Also, hearing which is "environmentally impaired" can be improved.
This approach uses the technique of testing the individual to determine what
enhancements are required or desired.
In this description, several specific circuits or devices are suggested.
These generally use the minimum means square spectral error filter criterion.
However, other types and designs of such circuits are contemplated. Such
a'ternative designs are within the knowledge of those skilled in the art. For
l example, the band pass filtered signal can be'frequency shifted if desired.
I However, any such modifications or alternatives which fall within the scope ofthis description are intended to be included therein as well.
Thus, the specific embodiments shown and described herein are intended to
be illustrative only, and are not intended to be limitative. Rather, the
scop of the inventSon i- limlted only by ~he claims appended hereto.
