EP0930801B1 - Schaltung und Verfahren zur adaptiven Unterdrückung einer akustischen Rückkopplung - Google Patents

Schaltung und Verfahren zur adaptiven Unterdrückung einer akustischen Rückkopplung Download PDF

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Publication number
EP0930801B1
EP0930801B1 EP98811273A EP98811273A EP0930801B1 EP 0930801 B1 EP0930801 B1 EP 0930801B1 EP 98811273 A EP98811273 A EP 98811273A EP 98811273 A EP98811273 A EP 98811273A EP 0930801 B1 EP0930801 B1 EP 0930801B1
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EP
European Patent Office
Prior art keywords
input signal
filter
correlation
echo
signal
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Expired - Lifetime
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EP98811273A
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German (de)
English (en)
French (fr)
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EP0930801A2 (de
EP0930801A3 (de
Inventor
Remo Leber
Arthur Schaub
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Bernafon AG
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Bernafon AG
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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
    • H04R25/45—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

  • the present invention relates to a circuit and a method for the adaptive suppression of an acoustic feedback according to the preambles of the independent claims. It is used, for example, in digital hearing aids.
  • acoustic feedback can occur between the loudspeaker or handset on the one hand and the microphone on the other hand.
  • the acoustic feedback causes unwanted distortions and leads in extreme cases to unstable behavior of the system, for example. An unpleasant whistling. Since the unstable operation is not acceptable, the signal gain of the signal processing part often has to be set smaller than effectively desired.
  • acoustic input signal is picked up and converted into a digital electrical signal. From this, an echo estimate is subtracted.
  • the echo-compensated signal is transformed with a necessary hearing correction into a digital output signal, converted into an analog electrical signal and emitted as an acoustic output signal.
  • the acoustic signal is deformed on its way back to the microphone according to a feedback characteristic and superimposed on an externally incident acoustic signal to a new acoustic input signal.
  • the fixed delays contained in the system are modeled and the unknown feedback characteristic is modeled.
  • a first approach involves the use of an artificial noise signal.
  • an artificial noise signal is, for example, from the European patent applications EP-415 677 .
  • the common feature of such systems is the use of an artificial noise signal to decorrelate the signals.
  • the noise signal is switched on only when needed instead of the output signal or continuously to Added output signal.
  • the disadvantage of these systems is the effort required for the control of the noise signal power such that the noise remains as inaudible and still a sufficiently good convergence speed can be achieved.
  • a third approach involves the use of adaptive decorrelation filters.
  • Such a system was, for example, in Mamadou Mboup et al., "Coupled Adaptive Prediction and System Identification: A Statistical Model and Transient Analysis", Proc. 1992 IEEE ICASSP, 4; 1-4, 1992 , described.
  • the feasible with this approach systems differ by the different arrangement and implementation of the decorrelation filter.
  • the disadvantage of the published system is the use of relatively slow transversal filter decorrelators, which due to their structure can not adapt very fast to the changing statistical properties of their input signals.
  • the coefficients of the two decorrelation filters are generally determined by decorrelation of the output signal reaching the loudspeaker or receiver. This is intended to make the convergence speed frequency-independent. A special weighting of the frequencies which are particularly critical for the feedback behavior with high gains in the signal processing path is thus not present.
  • an optimal convergence behavior with minimal, inaudible distortions and without additional signal delay should be achieved with the least possible effort.
  • the present invention belongs to the group of systems with adaptive decorrelation filters. It makes use of the knowledge that cross-member filter structures are particularly suitable for rapid decorrelation. Such cross-member filter structures are known from speech signal processing and are used there for linear prediction. Algorithms for the decorrelation of a signal by cross-link filter are known and can be taken from the literature, for example S. Thomas Alexander, "Adaptive Signal Processing", Springer-Verlag New York, 1986 ,
  • the present invention models the feedback path and adaptively follows its temporal changes by means of optimized tracking.
  • the fed back signal components are constantly removed from the input signal. This significantly increases the signal gain allowed for stable operation. This allows the use of higher reinforcements (eg in the case of severe hearing damage) or a pleasant, more open supply (eg in case of slight hearing damage).
  • the inventive circuit is used in an acoustic system with at least one microphone for generating an electrical input signal, at least one speaker or handset and an intermediate electronic signal processing part. It contains a filter for Modeling a feedback characteristic, an updating unit for calculating current coefficients for the filter, a subtractor for calculating an echo-compensated input signal by subtracting an echo estimate from a digital input signal supplied by the filter, a delay element for calculating a delayed output signal, and two adaptive cross-link decorrelation filters.
  • a first cross-sectional decorrelation filter is arranged to decorrelate the echo-canceled input signal
  • a second cross-sectional decorrelation filter is arranged to decorrelate the delayed output signal by means of coefficients derived from the first cross-sectional decorrelation filter.
  • the two cross-link decorrelation filters are configured to calculate their cross-member coefficients by means of adaptive decorrelation of the echo-canceled input signal.
  • the first decorrelation filter extracts from the echo-canceled signal the noise-like components contained therein.
  • the second decorrelation filter a cross-gate filter
  • the delayed output signal is converted into a transformed signal with the coefficients derived from the cross-divisor decorrelator.
  • the special feature of this arrangement is the interchanging of the cross-link decorrelator and the cross-link filter over the conventional arrangement, in which not the echo-canceled signal, but the delayed output signal is decorrelated.
  • the circuit according to the invention has the great advantage that the spectral maxima present in the hearing correction are preserved in the transformed signal. These maxima usually correspond to the most critical frequencies for the feedback, and these should certainly be taken into account in the updating of the filter coefficients with the correspondingly large weighting.
  • an electrical input signal is generated with at least one microphone, a feedback characteristic is modeled with a filter, current updating coefficients for the Calculates a filter, a subtracter calculates an echo-canceled input signal by subtracting an echo estimate provided by the filter from a digital input signal, and a delay element calculates a delayed output signal.
  • the echo-canceled input signal is decorrelated with a first cross-divisional decorrelation filter, and the delayed output signal is decorrelated with a second cross-divisional decorrelation filter by means of coefficients derived from the first cross-divisional decorrelation filter.
  • the cross-member coefficients of the two cross-member decorrelation filters are calculated by adaptive decorrelation of the echo-canceled input signal.
  • the present invention differs significantly from all previously published systems for suppressing the acoustic feedback. What is new are the special arrangement and realization of the blocks for decorrelation and normalization, the control of the forgetting factor and the step size factor, as well as the possibility of staggered updating in the combination according to the invention.
  • the present invention allows maximum convergence speeds with minimal distortion, since the updating of the filter coefficients takes place in terms of time and frequency mainly where the large amplifications in the auditory correction occur.
  • FIG. 1 A well-known system for the adaptive suppression of the acoustic feedback is in FIG. 1 shown.
  • An acoustic input signal a in (t) is picked up by a microphone 1 and initially converted into an electrical signal d (t).
  • a subsequent AD converter 2 determines a digital Input signal d n .
  • an echo estimate y n is subtracted in a subtractor 3.
  • the echo-compensated signal e n is transformed into a digital output signal u n with a correction 4 adaptable to the respective application, for example an individual hearing correction for a hearing impaired person.
  • the DA converter 5 performs a conversion into an electrical signal u (t), which is emitted via a loudspeaker or handset 6 as an acoustic output signal a out (t).
  • the acoustic output signal a out (t) is deformed on its way back to the microphone 1 in accordance with a feedback characteristic 7 characterized by an impulse response h ( ⁇ ) to a signal y (t) and superimposed on an externally incident acoustic signal s (t) (8 ).
  • the remaining components in the system are a delay element 9, a filter 10 and an updating unit 11.
  • the delay element 9 simulates the fixed delays contained in the system, resulting in a delayed signal x n .
  • the filter 10 models the unknown feedback characteristic.
  • the updating unit 11 the current coefficients w n for the filter are continuously calculated.
  • a variant of the LMS algorithm Least Mean Square
  • the well-known system does not suffice to achieve a distortion-free transmission in a realistic environment while at the same time satisfying convergence behavior.
  • the system can be improved if the updating unit works with decorrelated signals.
  • FIG. 2 shows a system which uses an artificial noise signal to decorrelate the signals.
  • a system is, for example, from the European patent applications EP-415 677 .
  • the artificial noise signal is generated in a noise generator 17 and added via a power control unit 18 to the digital output signal u n (19).
  • the artificial noise signal is also fed via a delay element 20 to the updating unit 11.
  • the noise signal is either switched on only if necessary instead of the output signal u n or continuously added to the output signal u n .
  • FIG. 3 shows a system which uses fixed orthogonal transformations to decorrelate the signals.
  • a system from Phonak AG was, for example, as a European patent application EP-585 976 released.
  • the echo-compensated signal e n and the output signal u n are transformed into the frequency domain via transformation units 21 and 22 and the echo estimate y n is recovered via an inverse transformation 23.
  • the filtering and updating of the coefficients in these systems does not take place directly in the time domain.
  • FIG. 4 shows a system which uses adaptive decorrelation filters 12, 13 to decorrelate the signals.
  • a system was, for example, in Mamadou Mboup et al., "Coupled Adaptive Prediction and System Identification: A Statistical Model and Transient Analysis", Proc. 1992 IEEE ICASSP, 4; 1-4, 1992 , described.
  • the echo-canceled signal e n and the delayed output signal x n are decorrelated by the adaptive decorrelation filters 12, 13.
  • the coefficients a n of the two decorrelation filters 12, 13 are calculated in block 13 by means of decorrelation of the delayed output signal x n .
  • FIG. 5 An embodiment of a system according to the invention is shown in FIG FIG. 5 shown.
  • the system according to the invention uses adaptive cross-member decorrelation filters, namely a cross-member decorrelator 12 and a cross-member filter 13 running parallel to it.
  • the cross-member filter structures known from speech signal processing prove to be particularly suitable , They are used there for linear prediction.
  • Algorithms for decorrelating a signal using cross-link filters are known.
  • the cross-correlator member 12 extracted from the echo-compensated signal e n is given by noise-like components e M n.
  • the cross member filter 13 with originating from the cross-over element de-correlator 12 coefficients k n the delayed output signal x n x M n converted into a transformed signal.
  • the special feature of this arrangement is the permutation of the two adaptive decorrelation filters 12 and 13 compared to the usual procedure, namely, not the echo-canceled signal e n , but the delayed signal x n is decorrelated.
  • the arrangement according to the invention has the great advantage that the spectral maxima present in the hearing correction 4 are maintained in the transformed signal x M n . These maxima usually correspond to the most critical frequencies for the feedback, and these should definitely be taken into account in the updating of the filter coefficients w n with the correspondingly large weighting.
  • the order of the two cross-link decorrelation filters 12, 13 is determined by a compromise between the desired degree of decorrelation and the associated computation effort.
  • M 2
  • a considerable improvement in the system behavior is again achieved by means of an upper boundary of the second cross-member coefficient k 2n .
  • This upper limit of the second cross-member coefficient has the consequence that pure sine tones are not completely decorrelated. This in turn has the great advantage that the whistling sounds occurring during unstable operation are compensated much faster.
  • the system according to the invention contains a control unit 14.
  • the control unit 14 continuously compares the power of the input signal d n with the power of the echo-compensated signal e n .
  • the ratio of the two powers determines which forgetting factor ⁇ n in the updating unit 11 Application comes. If the power of the echo-compensated signal is greater than the power of the input signal, this is almost always an indication that the echo estimate y n and thus the coefficients w n of the filter 10 are too large in terms of magnitude.
  • ⁇ n 1 is set.
  • the described control of the forgetting factor ⁇ n provides improved convergence behavior with rapid changes in the feedback path. An internal feedback generated temporarily by the system is recognized immediately and quickly adapted to the external feedback path.
  • the updating unit 11 includes a normalization unit 15 and a speed control unit 16.
  • the arrangement of the blocks described below is shown in FIG. 8 which illustrates a specification of the updating unit 11.
  • the normalization unit 15 allows the application of the NLMS algorithm (Normalized Least Mean Square). It calculates the power of the signal e M n .
  • the special feature of this arrangement is that the normalization with respect to e M n and not as usual with respect to x M n .
  • the convergence speed becomes dependent on the ratio of the powers of x M n and e M n .
  • This ratio is essentially given by the gain contained in the hearing correction 4.
  • the gain in the hearing correction is not constant in the general, non-linear case (eg compression method).
  • the convergence behavior of the adaptive filter 10 modeling the feedback characteristic 7 depends on the temporal behavior of the hearing correction 4, ie on the temporal course of its amplification and frequency response.
  • a fast adaptation of the coefficients w n occurs and in times of small amplification with uncritical feedback behavior a correspondingly slower adaptation takes place.
  • the updating takes place mainly in the times when it is actually necessary. This approach combines rapid convergence in the critical case with an approximately distortion-free processing in the uncritical case.
  • the speed control unit 16 provides a step factor ⁇ n for the NLMS algorithm.
  • the speed control unit 16 delivers values for ⁇ n beginning at the start value ⁇ max and within the first seconds after start-up gradually decreasing to the final value ⁇ min . This procedure allows a very fast convergence of the filter coefficients w n from zero to their nominal values after startup. The resulting initial signal distortions are less severe than the otherwise much longer-lasting feedback whistles.
  • the updating unit 11 can be designed so that only a certain small, cyclically changing part of the (N + 1) filter coefficients is updated at each discrete point in time. This considerably reduces the required computing effort. The system does not have to be made slower than it needs to be to prevent audible distortion anyway.
  • the microphone 1, the AD converter 2, the DA converter 5 and the handset 6 are considered ideal in the consideration.
  • the characteristics of the real acoustic and electrical transducers can be considered as part of the feedback characteristic 7.
  • T and f s denote the sampling period and the subscript n the discrete time.
  • the function f () stands for any nonlinear function of its arguments. It results from the selected procedure for correcting individual hearing loss.
  • the acoustic transmission path is modeled by means of the feedback characteristic 7 and an adder 8.
  • the operator * is to be understood as a convolution operator and h ( ⁇ ) stands for the impulse response of the feedback.
  • the externally incident signal is denoted by s (t).
  • the delay element 9 is in FIG. 6 and the following relationships apply.
  • the delay length L must be matched to the sum of the delays of the acoustic and electrical converters.
  • the filter 10 is in FIG. 7 and the following relationships apply. Underlined sizes mean the similar elements combined into vectors.
  • the factor r allows range selection, so that the filter coefficients are independent of the hearing correction 4 can always be kept in the range -1 ⁇ w kn ⁇ 1.
  • the filter order N must be matched to the length of the impulse response h ( ⁇ ).
  • the updating unit 11 is in FIG. 8 and the following relationships apply.
  • the formula is given in vector notation and element notation.
  • the updating unit 11 in turn includes the normalization unit 15 and the speed control unit 16.
  • the normalization unit 15 is shown in FIG. 9 and the following relationships apply.
  • the coefficients g and h determine the length of the time interval over which an averaging of the power of e M n takes place.
  • n n G ⁇ n n - 1 + H ⁇ e n M 2
  • G 63 / 64
  • the speed control unit 16 is in FIG. 10 and the following relationships apply.
  • the step size factor ⁇ n is gradually reduced by a factor of 0.5 starting from ⁇ max to ⁇ min .
  • the optimal values for ⁇ max and ⁇ min depend on the individual hearing correction (4).
  • the variable c n is used as a count variable.
  • ⁇ min ( c n 0 )
  • the cross-member decorrelator 12 is in FIG. 11 and the following relationships apply.
  • the quantities d i n and n i n must also be determined at each stage for tracking the coefficients k in .
  • the filter order M results from a compromise between the desired degree of decorrelation and the required computational effort.
  • the cross-member filter 13 is in FIG. 12 and the following relationships apply.
  • the control unit 14 is in FIG. 13 and the following relationships apply.
  • the forgetting factor ⁇ n results from the ratio of the two powers n d n and n e n .
  • the preferred embodiment can be easily programmed on a commercial signal processor or implemented in an integrated circuit. All variables must be suitably quantized and the operations optimized to the existing architecture blocks. Special attention is given to the treatment of the quadratic quantities (powers) and the division operations. Depending on the target system, there are optimized procedures for this. However, these are not in and of themselves subject of the present invention.

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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)
  • Filters That Use Time-Delay Elements (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
EP98811273A 1998-01-14 1998-12-30 Schaltung und Verfahren zur adaptiven Unterdrückung einer akustischen Rückkopplung Expired - Lifetime EP0930801B1 (de)

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CH6498 1998-01-14
CH6498 1998-01-14

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EP0930801A2 EP0930801A2 (de) 1999-07-21
EP0930801A3 EP0930801A3 (de) 2006-05-24
EP0930801B1 true EP0930801B1 (de) 2008-11-05

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AU (1) AU745946B2 (da)
DE (1) DE59814316D1 (da)
DK (1) DK0930801T3 (da)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2362687A2 (de) 2010-02-26 2011-08-31 Siemens Medical Instruments Pte. Ltd. Hörvorrichtung mit parallel betriebenen Rückkopplungsreduktionsfiltern und Verfahren

Families Citing this family (27)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6480610B1 (en) 1999-09-21 2002-11-12 Sonic Innovations, Inc. Subband acoustic feedback cancellation in hearing aids
EP1154674B1 (de) * 2000-02-02 2008-12-10 Bernafon AG Schaltung und Verfahren zur adaptiven Geräuschunterdrückung
US8374218B2 (en) * 2000-12-05 2013-02-12 Google Inc. Combining signals with a shuffled-hadamard function
US8385470B2 (en) * 2000-12-05 2013-02-26 Google Inc. Coding a signal with a shuffled-Hadamard function
US7545849B1 (en) 2003-03-28 2009-06-09 Google Inc. Signal spectrum spreading and combining system and method
US6829289B1 (en) * 2000-12-05 2004-12-07 Gossett And Gunter, Inc. Application of a pseudo-randomly shuffled hadamard function in a wireless CDMA system
US6982945B1 (en) 2001-01-26 2006-01-03 Google, Inc. Baseband direct sequence spread spectrum transceiver
US7453921B1 (en) * 2001-12-11 2008-11-18 Google Inc. LPC filter for removing periodic and quasi-periodic interference from spread spectrum signals
US7352833B2 (en) * 2002-11-18 2008-04-01 Google Inc. Method and system for temporal autocorrelation filtering
US7627129B2 (en) 2002-11-21 2009-12-01 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Apparatus and method for suppressing feedback
DE10254407B4 (de) * 2002-11-21 2006-01-26 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Unterdrücken einer Rückkopplung
AU2003236382B2 (en) 2003-08-20 2011-02-24 Phonak Ag Feedback suppression in sound signal processing using frequency transposition
AU2004201374B2 (en) 2004-04-01 2010-12-23 Phonak Ag Audio amplification apparatus
US7756276B2 (en) 2003-08-20 2010-07-13 Phonak Ag Audio amplification apparatus
US20050147258A1 (en) * 2003-12-24 2005-07-07 Ville Myllyla Method for adjusting adaptation control of adaptive interference canceller
EP1728091A4 (en) * 2003-12-24 2013-01-09 Nokia Corp EFFICIENT BEAM FORMING METHOD IMPLEMENTING AN ADDITIONAL NOISE SEPARATION FILTER
DE602004028502D1 (de) * 2004-12-22 2010-09-16 Televic Nv Verfahren und Anordnung für das Schätzen einer Raumimpulsantwort
JP2006197075A (ja) * 2005-01-12 2006-07-27 Yamaha Corp マイクロフォンおよび拡声装置
JP4215015B2 (ja) * 2005-03-18 2009-01-28 ヤマハ株式会社 ハウリングキャンセラ及びこれを備えた拡声装置
EP1793645A3 (en) 2005-11-09 2008-08-06 GPE International Limited Acoustical feedback suppression for audio amplification systems
US20070104335A1 (en) * 2005-11-09 2007-05-10 Gpe International Limited Acoustic feedback suppression for audio amplification systems
US8767972B2 (en) * 2006-08-16 2014-07-01 Apherma, Llc Auto-fit hearing aid and fitting process therefor
EP2165567B1 (en) * 2007-05-22 2010-10-06 Phonak AG Method for feedback cancelling in a hearing device and a hearing device
EP2071873B1 (en) * 2007-12-11 2017-05-03 Bernafon AG A hearing aid system comprising a matched filter and a measurement method
WO2015044915A1 (en) 2013-09-26 2015-04-02 Universidade Do Porto Acoustic feedback cancellation based on cesptral analysis
US9380387B2 (en) 2014-08-01 2016-06-28 Klipsch Group, Inc. Phase independent surround speaker
US10751524B2 (en) * 2017-06-15 2020-08-25 Cochlear Limited Interference suppression in tissue-stimulating prostheses

Family Cites Families (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB1520148A (en) * 1975-10-09 1978-08-02 Standard Telephones Cables Ltd Adaptive lattice filter
JPH02278926A (ja) * 1989-04-19 1990-11-15 Nec Corp Ｆｉｒ・ラティス混合形エコーキャンセラ
US5259033A (en) 1989-08-30 1993-11-02 Gn Danavox As Hearing aid having compensation for acoustic feedback
US5680467A (en) 1992-03-31 1997-10-21 Gn Danavox A/S Hearing aid compensating for acoustic feedback
DK170600B1 (da) * 1992-03-31 1995-11-06 Gn Danavox As Høreapparat med kompensation for akustisk tilbagekobling
DK169958B1 (da) 1992-10-20 1995-04-10 Gn Danavox As Høreapparat med kompensation for akustisk tilbagekobling
EP0585976A3 (en) * 1993-11-10 1994-06-01 Phonak Ag Hearing aid with cancellation of acoustic feedback
US5991418A (en) * 1996-12-17 1999-11-23 Texas Instruments Incorporated Off-line path modeling circuitry and method for off-line feedback path modeling and off-line secondary path modeling

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2362687A2 (de) 2010-02-26 2011-08-31 Siemens Medical Instruments Pte. Ltd. Hörvorrichtung mit parallel betriebenen Rückkopplungsreduktionsfiltern und Verfahren
DE102010009459A1 (de) * 2010-02-26 2011-09-01 Siemens Medical Instruments Pte. Ltd. Hörvorrichtung mit parallel betriebenen Rückkopplungsreduktionsfiltern und Verfahren
DE102010009459B4 (de) * 2010-02-26 2012-01-19 Siemens Medical Instruments Pte. Ltd. Hörvorrichtung mit parallel betriebenen Rückkopplungsreduktionsfiltern und Verfahren
US8737656B2 (en) 2010-02-26 2014-05-27 Siemens Medical Instruments Pte. Ltd. Hearing device with feedback-reduction filters operated in parallel, and method

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DK0930801T3 (da) 2009-02-23
EP0930801A2 (de) 1999-07-21
AU9826598A (en) 1999-08-05
AU745946B2 (en) 2002-04-11
EP0930801A3 (de) 2006-05-24
US6611600B1 (en) 2003-08-26
DE59814316D1 (de) 2008-12-18

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