EP0252205A2 - Appareil numérique pour améliorer l'audition - Google Patents

Appareil numérique pour améliorer l'audition Download PDF

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Publication number
EP0252205A2
EP0252205A2 EP87100741A EP87100741A EP0252205A2 EP 0252205 A2 EP0252205 A2 EP 0252205A2 EP 87100741 A EP87100741 A EP 87100741A EP 87100741 A EP87100741 A EP 87100741A EP 0252205 A2 EP0252205 A2 EP 0252205A2
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EP
European Patent Office
Prior art keywords
hearing
frequency
signal
signals
filter
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Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Ceased
Application number
EP87100741A
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German (de)
English (en)
Other versions
EP0252205A3 (fr
Inventor
Douglas M. Chabries
Robert H. Brey
Martin S. Robinette
Richard W. Christiansen
Gary R. Kenworthy
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Mark Antin
Original Assignee
Mark Antin
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Filing date
Publication date
Application filed by Mark Antin filed Critical Mark Antin
Publication of EP0252205A2 publication Critical patent/EP0252205A2/fr
Publication of EP0252205A3 publication Critical patent/EP0252205A3/fr
Ceased legal-status Critical Current

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/70—Adaptation of deaf aid to hearing loss, e.g. initial electronic fitting
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/45—Prevention of acoustic reaction, i.e. acoustic oscillatory feedback
    • H04R25/453—Prevention of acoustic reaction, i.e. acoustic oscillatory feedback electronically
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
    • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing

Definitions

  • noise supression techniques One type of hearing aid available on the market uses noise supression techniques. Hower, conventional filtering techniques generally are not considered to be ffective or adaquate for providing truly high fidelity frequency compensation which is desirable in hearing aids. Thus, results from implementation of these techniques often suffer from muffled sound outputs, and unacceptable noise and ringing problems.
  • a further problem in the conventional design of hearing aids is the inadequate treatment of background noise.
  • 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 vowels.
  • the user sacrifices speech intelligibility by simultaneously reducing the intensity of the lowe energy signals, e.g. sounds producd by consonants.
  • hearing aids which employ automatic gain control i.e. decrease gain as input level increases
  • background noise and vowels can have the same effect on the gain control, an abnormal relationship between spech sounds is introduced.
  • High frequency consonants for example, are not amplified sufficiently in the presence of backgroun noises thereby resulting in greatly reduced speech intelligence.
  • all sounds are amplified whereupon background noises greatly mask speech intelligibility.
  • 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 ability and, thus, the individual experiences a phenomenon known as a "cocktail party effect" in which allsounds 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 the conventional hearing aid.
  • 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 result is poor communication, frustration and fatique.
  • a standard hearing aid is of little or no advantageous consequence, for the reaons discussed above.
  • This invention is directed to a method and apparatus for improving the hearing capability of persons with sometype of impaired hearing, whether implicite or imposed.
  • the invention comprises a system which empirically detects the portions of a person's hearing which 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 improved hearing signal at the auditory nerve.
  • the invention does not merely amplify all sounds.
  • the 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, 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 are then operated upon separately. Appropriate suppression is achieved by adaptive filters, multiplication circuits of the like. Other operations such as taking the log 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.
  • 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 individfual hearing impairments.
  • Fig.l there is shown a typical graphical representation of a "normal" hearing pattern for the "average” human ear.
  • contours of equal loudness phons
  • the contours are numbered by the equal loudness correspondence with the intensity level at 1000 Hz.
  • the contours of equal loudness are, typically, spaced logarithmically and, hence, annotated in decibels (10 1og 10 ).
  • the human hearing system must account for the non-linearity.
  • contour 0 ist defined as the threshold of hearing. That ist, below this intensity the normal human ear does not perceive sound.
  • O dB and 1000 Hz a sound is just barely audible to the average person.
  • 50 dB and 1000 Hz the sound is well within the normal hearing range.
  • a 50 Hz signal normally is inaudible.
  • the upper contour is referred to as the threshold of pain discomfort. That is, the application of a signal of appropriate frequency 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.
  • LDL hearing aid fitting
  • a "loudness discomfort level” should be employed as an upper limit for hearing aid outputrather than a threshold of pain.
  • Fig.lA shows a graphic presentation of the sound pressure level (SPL) N2 frequency. This Figure also shows the mean and the range for comfortable (MCL) and uncomfortable listening levels (UCL) for pulsed narrow band noise. Subtracting the threshold levels from the upper range for the UCL, provides the dynamic range of hearing for "normal" hearing persons.
  • SPL sound pressure level
  • the dynamic range is between about 80 and 95 dB.
  • Impairment of hearing occurs when the threshold of hearing for an individual is, effectively, raised.
  • the dynamic range for that individual is reduced and possibly distorted.
  • 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 amplification of the signals, will be beneficial.
  • a uniform amplification thereof will amplify both desired frequencies (where a hearing 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.
  • 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 alteration in the shape of the thresholf hearing contour wherein certain frequency ranges are not received as well, or al all.
  • the human hearing system can be modeled as a non-linear process with measurable 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 signal-to-noise ration, yield improved speech intelligibility in the presence of noise including speech bablle, reduce or eliminate audio feedback and provide output between the threshold-of-hearing and the threshold-of-discomfort (LDL) contours for all frequencies.
  • the invention enhances loudness perception to the hearer.
  • the relationship of loudness in sones to loudness in phons for the normal ear is shown as the solid line 2A in Figure 2. This is a log/log plot where 40 phons equals 1 sone. Recruitment, an abnormally rapid growth in loudness, is represented by the dot-dashed line 2Bfor an individual with a 50 dB hearing loss at 1 KHz. That is, this individual cannot hear below 50 dB. However, the loudness grows rapidly until at 65 dB and 5 sones the loudness perception of the person is equal to that of a normal hearing system. This non-linearity must be taken into account for the hearing impaired listener.
  • FIG. 3 there is shown a functional block diagram which is representative of a non-linear model of the hearing operation of the human hearing system 300.
  • sound is provided by a typical source 303 and received in the ear apparatus.
  • the ear operates as a frequency transducer 301 which separates the incoming sound signal into a plurality of band pass output signals A.
  • 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 selectively reject background, or noise, signals and concentrate on the desired signals.
  • the signals B from the transfer function 302 are provided to the log circuit 303 which performs a logarithmic function thereon.
  • the output C of the log function 303 is supplied to the recruitment function 304 which, effectively, scales the supplied signals as a function of frequency to producee an output with a dynamic range which fits between the threshold-of-hearing and the threshold-of-discomfort (i.e., the dynamic range of the ear) for all hearing range frequencies.
  • the output D of the recruitment function 304 is supplied to the clipping or saturation function 305 which has the effect of cutting off extremely low and high amplitudes by saturating.
  • the output E of the clipping function 305 is provided in what is referred to as the "perceptual space" 306.
  • This perceptual space is, for purposes of this discussion, defined as the signal space at the input ends of auditory nerve.
  • the effect that is produced by the hearing system is, essentially, the mapping of signals to the auditory nerve input, which will then simulate nerve firings or the like, which can then be detected as appropriate sounds.
  • the hearing operation and the impairment thereof is a function of the operation of one or more of the functions shown and described in the "dual" of the human hearing system shown in Figure 3.
  • the sensitivity function 302 the log function 303, the recruitment function 304, or the clipping function 305 is, in some way defective, a portion of the band pass signals supplied by the frequency transformation function 301 are lost, diminished, enhanced, or the like. This los can be produced at signal level A, B, C, D 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.
  • a hearing enhancement device 100 can be interposed between the sound source 308 and the mechanism 300 which represents the human hearing system.
  • This hearing enhancement device 100 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.
  • the hearing aid device 100 is used in an attempt to compensate for any deficiencies in the actual hearing mechanism 300.
  • the individual is tested, in an 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.
  • FIG. 4 there is shown a schematic representation of a system incorporating the hearing aid of this invention.
  • 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.
  • these filters can be symetric band pass filters evenly spaced across the bandwidth of the input signal.
  • 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.
  • the sensitivity circiut 409 is the inverse of 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 base 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.
  • digital signal processing techniques for feedback 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 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.
  • the transmultiplexer is, essentially, comprised of the five component portions including the input pre-filtering stage 501, the time-to-frequency transforms (FFT) 502, the processing blocks in the transform space 503, the frequency-to-time transforms (inverse FFT) 504, and the output post-filtering stage 505.
  • the processing blocks include a noise supression stage 506, a feedback supression stage 507, a frequency compensation stage 508 and a recruitment stage 509.
  • the transmultiplexer 500 operates on the basis of an algorithm which transforms a time signal to its frequency representation at stages 501 and 502, allows independent processing between frequency bins in the transform space 503, and then transforms the frequency representation back into a time signal (stages 504 and 505).
  • the transmultiplexer is used to maximize the homomorphic processing potential in the transform space 503 by assuring that the bins in the transform space are essentially independent.
  • an FFT is a computationally efficient algorithm for obtaining the frequency representation of atime signal.
  • the output of an N point FFT is N frequency bins, each approximating the amplitude of the time signal in that frequency range.
  • the value in a particular frequency bin is not a function of the energy at that frequency alone, but, rather, there is a significant interaction between the actual energies in several adjacent bins. Inasmuch as the values in the bins are not independent, one bin cannot be scaled without affecting other frequency bins when the inverse FFT function is performed.
  • the transmultiplexer algorithm uses two overlapped FFT's, as well as input and output filtering, to decrease dependence between frequency bins. The frequency bins do not overlap significantly with bins adjacent thereto.
  • the inputs to each FFT 502A and 502B are the outputs of two separate input filter banks 501A and 501B, respectively.
  • the input filter banks have the same coefficients but the input 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.
  • 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 window function noted above.
  • the output signal from filter bank 505A is passed through delay 511 and delayed by N/2 and then added to the output signal of filter bank 505B at summing junction 512 to yield the processed transmultiplexer output.
  • the inputs to the two output filter banks 505A and 505B are the outputs of the two overlapped inverse FFTs 504A and 504B.
  • the algorithms of F FT 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.
  • noise suppressor 506 comprises a bank of adaptive filter 601.
  • Each of the adaptive filters includes a FIR-filter 602 with feedback 603. There is one filter per bin thereby realizing the symmetry savings noted above.
  • Each filter 601 may include a different p forming a vector ⁇ when considering all filters in the filter bank. The vector permits control of the adaptation times in the frequency bins. If noise suppression is employed at the input to band pass filter 401 or processing circuit 403, in the system of Figure 4, then the ⁇ for each frequency region will be different to allow equal adaptation times.
  • the bulk delay 603 incorporates a delay time Z - ⁇ and is used to decorrelate the "primary" input to the filter with the "desired” response.
  • the delay time, A in this embodiment is equivalent to ⁇ x N/2 samples. This permits noise suppression in the adaptive filter.
  • FIG. 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: where a, b, and c are constant filter coefficients, the subscript j indicates sample j and Z is the standard notation for a unit sample delay. These coefficients are selected as noted above.
  • the coefficients for the filters are samples from a window function which modifies the input signal so the bins in the sample space will not overlap. Any window 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 example, 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.
  • 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, and requires transforming this delayed signal into the transform domain.
  • the delayed input 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 A 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.
  • each frequency bin is multiplied by some attenuation factor A k (m).
  • This attenuation factor is determined from the smoothed power (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 equation: and where k denotes the frequency bin, M denotes the sample number, N2 is the estimated noise power, X 2 k ist the smoothed power, and P k 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 A in Figure 7 is transformed into the transform space 503.
  • the delay time is equivalent to A 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.
  • FIG 8. Another method of noise supression 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 supression is also called spectral subtraction.
  • 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.
  • the bulk feeddback 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.
  • the output of the feedback suppressor is defined by the Error signal.
  • FIG 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.
  • 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 "table lookup", with one table for each bin.
  • the table contents are scale factors, much like the frequency equalization scale factors, 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.
  • the recruitment device 509 for example, will produce the appropriate "output”.
  • This output will be appropriate to enhance the individual's hearing within the prescribed dynamic range.
  • the actual hearing capability of the user is enhanced and optimized.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Filters That Use Time-Delay Elements (AREA)
EP87100741A 1986-01-21 1987-01-20 Appareil numérique pour améliorer l'audition Ceased EP0252205A3 (fr)

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US82063286A 1986-01-21 1986-01-21
US820632 1986-01-21

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EP0252205A2 true EP0252205A2 (fr) 1988-01-13
EP0252205A3 EP0252205A3 (fr) 1989-09-27

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EP87100741A Ceased EP0252205A3 (fr) 1986-01-21 1987-01-20 Appareil numérique pour améliorer l'audition

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EP (1) EP0252205A3 (fr)
JP (1) JPS62224200A (fr)
AU (1) AU596633B2 (fr)
CA (1) CA1284529C (fr)
DK (1) DK33587A (fr)

Cited By (9)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4852175A (en) * 1988-02-03 1989-07-25 Siemens Hearing Instr Inc Hearing aid signal-processing system
EP0339819A2 (fr) * 1988-04-11 1989-11-02 Central Institute For The Deaf Filtre électronique
EP0535425A2 (fr) * 1991-10-03 1993-04-07 Ascom Audiosys Ag Procédé d'amplification de signaux acoustiques pour les malentendants et dispositif pour la réalisation du procédé
EP0585976A2 (fr) * 1993-11-10 1994-03-09 Phonak Ag Prothèse auditive avec suppression du couplage acoustique
US5357251A (en) * 1988-03-23 1994-10-18 Central Institute For The Deaf Electronic filters, signal conversion apparatus, hearing aids and methods
EP0661905A3 (fr) * 1995-03-13 1995-10-04 Phonak Ag Procédé d'adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive.
US5475759A (en) * 1988-03-23 1995-12-12 Central Institute For The Deaf Electronic filters, hearing aids and methods
DE19703228A1 (de) * 1997-01-29 1998-07-30 Siemens Audiologische Technik Verfahren zur Verstärkung von Eingangssignalen eines Hörgerätes sowie Schaltung zur Durchführung des Verfahrens
US6327366B1 (en) 1996-05-01 2001-12-04 Phonak Ag Method for the adjustment of a hearing device, apparatus to do it and a hearing device

Families Citing this family (2)

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US5027410A (en) * 1988-11-10 1991-06-25 Wisconsin Alumni Research Foundation Adaptive, programmable signal processing and filtering for hearing aids
CA2462463A1 (fr) 2004-03-30 2005-09-30 Dspfactory Ltd. Methode et systeme de reduction des effets secondaires audibles de la consommation dynamique de courant

Citations (1)

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EP0071845A2 (fr) * 1981-08-06 1983-02-16 Siemens Aktiengesellschaft Appareil pour la compensation des carences d'audition

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SE428167B (sv) * 1981-04-16 1983-06-06 Mangold Stephan Programmerbar signalbehandlingsanordning, huvudsakligen avsedd for personer med nedsatt horsel
DK546581A (da) * 1981-12-10 1983-06-11 Danavox As Fremgangsmaade til tilpasning af overfoeringsfunktionen i et hoereapparat til forskellige hoeredefekter samt hoereapparat til udoevelse af fremgangsmaaden
AU569591B2 (en) * 1983-10-25 1988-02-11 Australian Hearing Services Hearing aid amplification method and apparatus

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EP0071845A2 (fr) * 1981-08-06 1983-02-16 Siemens Aktiengesellschaft Appareil pour la compensation des carences d'audition

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Title
PROCEEDINGS OF THE SEVENTH ANNUAL CONFERENCE OF THE IEEE/ENGINEERING IN MEDICINE AND BIOLOGY SOCIETY, Chicago, 27th - 30th September 1985, vol. 2, pages 1103-1108, IEEE; D. GRAUPE et al.: "Self adaptive filtering of background noise in hearing aids" *

Cited By (17)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0326905A1 (fr) * 1988-02-03 1989-08-09 Siemens Aktiengesellschaft Système d'élaboration de signaux pour prothèse auditive
US4852175A (en) * 1988-02-03 1989-07-25 Siemens Hearing Instr Inc Hearing aid signal-processing system
US5357251A (en) * 1988-03-23 1994-10-18 Central Institute For The Deaf Electronic filters, signal conversion apparatus, hearing aids and methods
US5475759A (en) * 1988-03-23 1995-12-12 Central Institute For The Deaf Electronic filters, hearing aids and methods
EP0339819A2 (fr) * 1988-04-11 1989-11-02 Central Institute For The Deaf Filtre électronique
EP0339819A3 (fr) * 1988-04-11 1991-01-30 Central Institute For The Deaf Filtre électronique
EP0535425A2 (fr) * 1991-10-03 1993-04-07 Ascom Audiosys Ag Procédé d'amplification de signaux acoustiques pour les malentendants et dispositif pour la réalisation du procédé
EP0535425A3 (en) * 1991-10-03 1993-11-10 Ascom Audiosys Ag Method for amplifying an acoustic signal for the hard of hearing and device for carrying out the method
EP0585976A3 (en) * 1993-11-10 1994-06-01 Phonak Ag Hearing aid with cancellation of acoustic feedback
EP0585976A2 (fr) * 1993-11-10 1994-03-09 Phonak Ag Prothèse auditive avec suppression du couplage acoustique
EP0656737A1 (fr) * 1993-11-10 1995-06-07 Phonak Ag Prothèse auditive avec suppression du couplage acoustique
EP0661905A3 (fr) * 1995-03-13 1995-10-04 Phonak Ag Procédé d'adaptation de prothèse auditive, dispositif à cet effet et prothèse auditive.
US6327366B1 (en) 1996-05-01 2001-12-04 Phonak Ag Method for the adjustment of a hearing device, apparatus to do it and a hearing device
US7231055B2 (en) 1996-05-01 2007-06-12 Phonak Ag Method for the adjustment of a hearing device, apparatus to do it and a hearing device
DE19703228A1 (de) * 1997-01-29 1998-07-30 Siemens Audiologische Technik Verfahren zur Verstärkung von Eingangssignalen eines Hörgerätes sowie Schaltung zur Durchführung des Verfahrens
US6198830B1 (en) 1997-01-29 2001-03-06 Siemens Audiologische Technik Gmbh Method and circuit for the amplification of input signals of a hearing aid
DE19703228B4 (de) * 1997-01-29 2006-08-03 Siemens Audiologische Technik Gmbh Verfahren zur Verstärkung von Eingangssignalen eines Hörgerätes sowie Schaltung zur Durchführung des Verfahrens

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DK33587A (da) 1987-07-22
DK33587D0 (da) 1987-01-21
EP0252205A3 (fr) 1989-09-27
AU6767187A (en) 1987-07-23
AU596633B2 (en) 1990-05-10
JPS62224200A (ja) 1987-10-02
CA1284529C (fr) 1991-05-28

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