EP0930801A2 - Circuit et procédé pour la suppression adaptative de la réaction acoustique - Google Patents

Circuit et procédé pour la suppression adaptative de la réaction acoustique Download PDF

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Publication number
EP0930801A2
EP0930801A2 EP98811273A EP98811273A EP0930801A2 EP 0930801 A2 EP0930801 A2 EP 0930801A2 EP 98811273 A EP98811273 A EP 98811273A EP 98811273 A EP98811273 A EP 98811273A EP 0930801 A2 EP0930801 A2 EP 0930801A2
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European Patent Office
Prior art keywords
filter
decorrelation
input signal
echo
cross
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Granted
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EP98811273A
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German (de)
English (en)
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EP0930801A3 (fr
EP0930801B1 (fr
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 adaptive Suppressing acoustic feedback according to the generic terms of independent claims. It is used, for example, in digital hearing aids Commitment.
  • a loudspeaker or handset and an intermediate electronic signal processing part may an acoustic feedback between loudspeaker or handset on the one hand and Microphone come on the other hand.
  • the acoustic feedback caused unwanted distortion and in extreme cases leads to unstable behavior of the Systems, for example an uncomfortable whistle. Because the unstable operation is not is acceptable, the signal amplification of the signal processing part often has to be lower be set as effectively desired.
  • the suppression of acoustic feedback in digital hearing aids can basically be approached with different approaches.
  • the best Results are currently achieved using the adaptive filtering method.
  • an acoustic input signal is recorded and integrated into a digital electrical signal converted. This will be an echo estimate deducted.
  • the echo-compensated signal comes with a necessary hearing correction transformed into a digital output signal, into an analog electrical signal converted and emitted as an acoustic output signal.
  • the acoustic On its way back to the microphone, the signal becomes corresponding to one Feedback characteristic deformed and one from outside acoustic signal superimposed on a new acoustic input signal.
  • For Calculation of the echo estimate will be the fixed ones included in the system Replicated delays and the unknown feedback characteristic modeled.
  • a first approach involves the use of an artificial noise signal.
  • a such a system is known, for example, from European patent applications EP-415 677, EP-634 084 and EP-671 114 from GN Danavox AS are known.
  • the common The property of such systems is the use of an artificial one Noise signal for decorrelation of the signals.
  • the noise signal is either only if necessary, switched on instead of the output signal or ongoing to Output signal added.
  • the disadvantage of these systems is the effort required for the control of the noise signal power in such a way that the noise is as possible remains inaudible and still has a sufficiently good speed of convergence can be achieved.
  • a third approach involves the use of adaptive decorrelation filters.
  • Such a system has been described, 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.
  • the one with this approach feasible systems differ in the different arrangement and Realization of the decorrelation filter.
  • the disadvantage of the published system is in the use of relatively slow transversal filter decorrelators due to their structure, the changing statistical is not particularly fast Can adjust properties of their input signals.
  • the coefficients of the two decorrelation filters are generally by decorrelation of the to Loudspeaker or receiver output signal determined. So that should Convergence speed can be made frequency independent. A special weighting of those that are particularly critical for the feedback behavior Frequencies with high gains in the signal processing path are not available.
  • the present invention belongs to the group of systems with adaptive Decorrelation filters. It takes advantage of the knowledge that cross-link filter structures are particularly suitable for fast decorrelation. Such Cross-link filter structures are known from speech signal processing and are used there for linear prediction. Algorithms for decorrelation of a signal using a cross-link filter are known and can be found in the specialist literature are taken, for example, from S. Thomas Alexander, "Adaptive Signal Processing ", Springer-Verlag New York, 1986.
  • the present invention models and follows the feedback path Adaptive changes in time using optimized tracking.
  • the Feedback signal components are continuously removed from the input signal.
  • the signal amplification permissible for stable operation thus becomes essential elevated. This enables the use of higher reinforcements (e.g. heavy ones Hearing damage) or a more pleasant open care (e.g. with light Hearing loss).
  • the circuit according to the invention comes with an acoustic system at least one microphone for generating an electrical input signal, at least one speaker or handset and one in between electronic signal processing part for use. It contains a filter for Modeling of a feedback characteristic, an update unit for Calculation of current coefficients for the filter, a subtractor for Calculation of an echo-compensated input signal by subtracting one echo estimate from a digital input signal provided by the filter Delay element for calculating a delayed output signal and two adaptive cross-link decorrelation filters.
  • a first cross-link decorrelation filter is for decorrelation of the echo-compensated input signal arranged
  • a second cross-link decorrelation filter is for decorrelation of the delayed output signal by means of the first cross-member decorrelation filter originating coefficients arranged.
  • the two cross-link decorrelation filters are used to calculate their cross-link coefficients adaptive decorrelation of the echo-compensated input signal configured.
  • the first decorrelation filter a cross-link decorrelator, extracted from the echo-compensated signal the noise-like components contained therein.
  • a cross-section filter with the the delayed coefficient from the cross-link decorrelator Output signal converted into a transformed signal.
  • the special thing about this arrangement is the exchange of the cross-link decorrelator and the Cross-link filter compared to the usual arrangement, in which not that echo-compensated signal, but decorrelated the delayed output signal becomes.
  • the circuit according to the invention has the great advantage that the Preservation of existing spectral maxima in the transformed signal stay. These maxima mostly correspond to those for the feedback most critical frequencies, and these should be used when updating the Filter coefficients with the correspondingly large weighting be taken into account.
  • the inventive method for adaptive suppression of the Acoustic feedback becomes electrical with at least one microphone Input signal generated with a filter a feedback characteristic modeled with an update unit current coefficients for the Filter calculated, with a subtractor is an echo-compensated Input signal by subtracting an echo estimate from the filter calculated with a digital input signal, and with a delay element a delayed output signal is calculated.
  • a first cross-link decorrelation filter the echo-compensated input signal is decorrelated, and with the delayed output signal becomes a second cross-member decorrelation filter by means of coefficients originating from the first cross-link decorrelation filter decorrelated.
  • the cross-link coefficients of the two cross-link decorrelation filters using adaptive decorrelation of the echo-compensated input signal calculated.
  • the present invention differs significantly from all of them so far published systems for the suppression of acoustic feedback.
  • the present invention allows maximum Convergence speeds with minimal distortion as the update the filter coefficients mainly take place there in terms of time and frequency, where the big gains in hearing correction occur.
  • FIG. 1 A generally known system for adaptive suppression of acoustic feedback is shown in FIG. 1 .
  • 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 therefrom.
  • An echo estimate y n is subtracted from this in a subtractor 3.
  • the echo-compensated signal e n is transformed into a digital output signal u n with a correction 4 that can be adapted to the respective application, for example an individual hearing correction for a hearing impaired person.
  • the DA converter 5 carries out a conversion into an electrical signal u (t), which is emitted via a loudspeaker or receiver 6 as an acoustic output signal a out (t).
  • the acoustic output signal a out (t) is deformed into a signal y (t) in accordance with a feedback characteristic 7 characterized by an impulse response h (t) and is superimposed on an acoustic signal s (t) incident from the outside (8 ).
  • the remaining components in the system are a delay element 9, a filter 10 and an update unit 11.
  • the delay element 9 simulates the fixed delays contained in the system, which results in a delayed signal x n .
  • the filter 10 models the unknown feedback characteristic.
  • the current coefficients w n for the filter are continuously calculated in the update unit 11.
  • a variant of the LMS algorithm Least Mean Square
  • the generally known system is sufficient because of the not negligible Autocorrelation function of real acoustic signals s (t) not to be realistic Environment a low-distortion transmission with satisfactory at the same time To achieve convergence behavior.
  • the system can be improved if the Update unit works with decorrelated signals.
  • FIG. 2 shows a system which uses an artificial noise signal to decorrelate the signals.
  • a system is known, for example, from European patent applications EP-415 677, EP-634 084 and EP-671 114 from GN Danavox AS.
  • the artificial noise signal is generated in a noise generator 17 and added to the digital output signal u n via a power control unit 18 (FIG. 19).
  • the artificial noise signal is also fed to the update unit 11 via a delay element 20.
  • the noise signal is either switched on only when required instead of the output signal u n or is continuously added to the output signal u n .
  • FIG. 3 shows a system which uses fixed orthogonal transformations for the decorrelation of the signals.
  • a system from Phonak AG was published, for example, as a European patent application EP-585 976.
  • the echo-compensated signal e n and the output signal u n are transformed into the frequency range 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 is not carried out directly in the time domain in these systems.
  • FIG. 4 shows a system which uses adaptive decorrelation filters 12, 13 for the decorrelation of the signals.
  • adaptive decorrelation filters 12, 13 for the decorrelation of the signals.
  • Such a system has been described, 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.
  • the echo-compensated 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 exemplary embodiment of a system according to the invention is shown in FIG. 5 .
  • the system according to the invention uses adaptive cross-link decorrelation filters, namely a cross-link decorrelator 12 and a cross-link filter 13 running in parallel therewith.
  • the cross-link filter structures known from speech signal processing have proven to be particularly suitable for fast decorrelation . They are used there for linear prediction. Algorithms for the decorrelation of a signal using a cross-link filter are known.
  • the cross-correlator member 12 extracted from the echo-canceled signal e noise-like component given by e n M n therein.
  • the special feature of this arrangement is the interchanging of the two adaptive decorrelation filters 12 and 13 compared to the usual procedure, in which the delayed signal x n is decorrelated rather than the echo-compensated signal e n .
  • the arrangement according to the invention has the great advantage that the spectral maxima present in the hearing correction 4 are retained in the transformed signal x M n . These maxima mostly correspond to the most critical frequencies for the feedback, and these should be taken into account with the correspondingly large weighting when updating the filter coefficients w n .
  • the order of the two cross slide decorrelation filters 12, 13 is determined from a compromise between the desired degree of decorrelation and the computational effort involved.
  • This upper limit of the second cross-link coefficient means that pure sine tones are not completely decorrelated. This in turn has the great advantage that the whistling tones that occur during unstable operation are compensated for much more quickly.
  • 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 is used in the update unit 11. 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 amount.
  • ⁇ n 1 is set.
  • the described control of the forgetting factor ⁇ n provides an 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 update unit 11 contains a standardization unit 15 and a speed control unit 16.
  • the arrangement of the blocks described below can be seen from FIG. 8, which represents a more precise description of the update unit 11.
  • the normalization unit 15 enables the NLMS (Normalized Least Mean Square) algorithm to be used. It calculates the power of the signal e M n .
  • the special thing about this arrangement is that the standardization is done with respect to e M n and not with x M n as usual. The rate of convergence thus becomes dependent on the ratio of the powers of x M n and e M n . This ratio is essentially given by the amplification contained in the hearing correction 4.
  • the gain in the hearing correction is generally not constant over time in the non-linear case (e.g. compression method).
  • the convergence behavior of the adaptive filter 10 modeling the feedback characteristic 7 therefore depends on the temporal behavior of the hearing correction 4, ie on the temporal course of its amplification and frequency response.
  • the coefficients w n are rapidly adapted and in times of small amplification with non-critical feedback behavior, a correspondingly slower adaptation takes place.
  • the update takes place mainly in the times when it is actually necessary. This procedure combines rapid convergence in the critical case with almost distortion-free processing in the uncritical case.
  • the speed control unit 16 supplies a step size factor ⁇ n for the NLMS algorithm.
  • the speed control unit 16 supplies values for ⁇ n starting with the standard value ⁇ max and gradually decreasing within the first seconds after starting up to the final value ⁇ min . After starting, this procedure allows the filter coefficients w n to converge very quickly from zero to their target values. The resulting initial signal distortion is less serious than the otherwise much longer feedback whistle.
  • the updating unit 11 can be designed so that each discrete Time only a certain small, cyclically changing part of the (N + 1) Filter coefficients is updated. This reduces the computing effort required considerably. The system does not have to be slowed down, than it has to be to prevent audible distortion anyway.
  • 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 signal coming in from outside is denoted by s (t).
  • the delay element 9 is shown in Figure 6 and the following relationships apply.
  • the delay length L must be matched to the sum of the delays of the acoustic and electrical transducers.
  • Filter 10 is shown in Figure 7 and the following relationships apply. Underlined sizes mean the similar elements combined into vectors.
  • the factor r allows a range to be selected so that the filter coefficients can always be kept in the range -1 ⁇ w kn ⁇ 1 regardless of the hearing correction 4.
  • the filter order N must be matched to the length of the impulse response h ( ⁇ ).
  • the update unit 11 is shown in FIG. 8 , and the following relationships apply.
  • the formula is given in vector notation and in element notation.
  • the updating unit 11 in turn contains 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 the power of e M n is averaged.
  • the speed control unit 16 is shown in Figure 10 and the following relationships apply.
  • the step size factor ⁇ n is gradually reduced from ⁇ max by a factor of 0.5 to ⁇ min .
  • the optimal values for ⁇ max and ß min depend on the individual hearing correction (4).
  • the variable c n is used as a counter variable.
  • Cross-link decorrelator 12 is shown in Figure 11 and the following relationships apply.
  • the quantities d i n and n i n must also be determined at each level for the tracking of the coefficients k in .
  • the filter order M results from a compromise between the desired degree of decorrelation and the required computing effort.
  • the cross-link filter 13 is shown in Figure 12 and the following relationships apply.
  • the control unit 14 is shown 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 . There is a hysteresis in the middle area.
  • the preferred embodiment can easily be on a commercial Signal processor programmed or implemented in an integrated circuit become. To do this, all variables must be appropriately quantized and the operations based on the existing architectural blocks are optimized. A special Attention is paid to the treatment of square sizes (services) and the division operations. Depending on the target system, there are optimized ones Procedures. But in and of themselves these are not the subject of 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 Circuit et procédé pour la suppression adaptative de la réaction acoustique Expired - Lifetime EP0930801B1 (fr)

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

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EP0930801A3 EP0930801A3 (fr) 2006-05-24
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AU (1) AU745946B2 (fr)
DE (1) DE59814316D1 (fr)
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Cited By (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2001022775A2 (fr) * 1999-09-20 2001-03-29 Sonic Innovations, Inc. Suppression de l'effet larsen de sous-bandes dans des protheses auditives
EP1154674A2 (fr) * 2000-02-02 2001-11-14 Bernafon AG Circuit et méthode pour la suppression adaptive du bruit
DE10254407A1 (de) * 2002-11-21 2004-06-17 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Unterdrücken einer Rückkopplung
US7627129B2 (en) 2002-11-21 2009-12-01 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Apparatus and method for suppressing feedback
US7756276B2 (en) 2003-08-20 2010-07-13 Phonak Ag Audio amplification apparatus
US7778426B2 (en) 2003-08-20 2010-08-17 Phonak Ag Feedback suppression in sound signal processing using frequency translation
US8351626B2 (en) 2004-04-01 2013-01-08 Phonak Ag Audio amplification apparatus
US9380387B2 (en) 2014-08-01 2016-06-28 Klipsch Group, Inc. Phase independent surround speaker

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US7545849B1 (en) 2003-03-28 2009-06-09 Google Inc. Signal spectrum spreading and combining system and method
US8374218B2 (en) 2000-12-05 2013-02-12 Google Inc. Combining signals with a shuffled-hadamard function
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
US8385470B2 (en) * 2000-12-05 2013-02-26 Google Inc. Coding a signal with a shuffled-Hadamard function
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
US20050147258A1 (en) * 2003-12-24 2005-07-07 Ville Myllyla Method for adjusting adaptation control of adaptive interference canceller
WO2005065012A2 (fr) * 2003-12-24 2005-07-21 Nokia Corporation Procede de formation de faisceau efficace mettant en application un filtre de separation de bruit complementaire
ATE476826T1 (de) * 2004-12-22 2010-08-15 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
US20070104335A1 (en) * 2005-11-09 2007-05-10 Gpe International Limited Acoustic feedback suppression for audio amplification systems
EP1793645A3 (fr) 2005-11-09 2008-08-06 GPE International Limited Suppression de la rétroaction acoustique pour les systèmes de amplification audio
US8767972B2 (en) * 2006-08-16 2014-07-01 Apherma, Llc Auto-fit hearing aid and fitting process therefor
WO2007125132A2 (fr) * 2007-05-22 2007-11-08 Phonak Ag Procédé d'annulation du retour dans un appareil auditif et appareil auditif ainsi obtenu
EP2475192A3 (fr) * 2007-12-11 2015-04-01 Bernafon AG Système d'assistance auditive comprenant un filtre adapté et procédé de mesure
DE102010009459B4 (de) 2010-02-26 2012-01-19 Siemens Medical Instruments Pte. Ltd. Hörvorrichtung mit parallel betriebenen Rückkopplungsreduktionsfiltern und Verfahren
WO2015044915A1 (fr) 2013-09-26 2015-04-02 Universidade Do Porto Annulation de réaction acoustique sur la base d'une analyse cepstrale
US10751524B2 (en) * 2017-06-15 2020-08-25 Cochlear Limited Interference suppression in tissue-stimulating prostheses

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Cited By (13)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2001022775A2 (fr) * 1999-09-20 2001-03-29 Sonic Innovations, Inc. Suppression de l'effet larsen de sous-bandes dans des protheses auditives
US7020297B2 (en) 1999-09-21 2006-03-28 Sonic Innovations, Inc. Subband acoustic feedback cancellation in hearing aids
WO2001022775A3 (fr) * 1999-09-21 2001-12-06 Sonic Innovations Inc Suppression de l'effet larsen de sous-bandes dans des protheses auditives
US6480610B1 (en) 1999-09-21 2002-11-12 Sonic Innovations, Inc. Subband acoustic feedback cancellation in hearing aids
EP1154674A2 (fr) * 2000-02-02 2001-11-14 Bernafon AG Circuit et méthode pour la suppression adaptive du bruit
EP1154674A3 (fr) * 2000-02-02 2007-03-21 Bernafon AG Circuit et méthode pour la suppression adaptive du bruit
DE10254407A1 (de) * 2002-11-21 2004-06-17 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Unterdrücken einer Rückkopplung
DE10254407B4 (de) * 2002-11-21 2006-01-26 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Unterdrücken einer Rückkopplung
US7627129B2 (en) 2002-11-21 2009-12-01 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Apparatus and method for suppressing feedback
US7756276B2 (en) 2003-08-20 2010-07-13 Phonak Ag Audio amplification apparatus
US7778426B2 (en) 2003-08-20 2010-08-17 Phonak Ag Feedback suppression in sound signal processing using frequency translation
US8351626B2 (en) 2004-04-01 2013-01-08 Phonak Ag Audio amplification apparatus
US9380387B2 (en) 2014-08-01 2016-06-28 Klipsch Group, Inc. Phase independent surround speaker

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Publication number Publication date
US6611600B1 (en) 2003-08-26
AU745946B2 (en) 2002-04-11
EP0930801A3 (fr) 2006-05-24
DK0930801T3 (da) 2009-02-23
AU9826598A (en) 1999-08-05
DE59814316D1 (de) 2008-12-18
EP0930801B1 (fr) 2008-11-05

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