EP3065417A1 - Procede de suppression d'un bruit parasite dans un systeme acoustique - Google Patents

Procede de suppression d'un bruit parasite dans un systeme acoustique Download PDF

Info

Publication number
EP3065417A1
EP3065417A1 EP16151092.0A EP16151092A EP3065417A1 EP 3065417 A1 EP3065417 A1 EP 3065417A1 EP 16151092 A EP16151092 A EP 16151092A EP 3065417 A1 EP3065417 A1 EP 3065417A1
Authority
EP
European Patent Office
Prior art keywords
signal
input
microphone
filter
output signal
Prior art date
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.)
Granted
Application number
EP16151092.0A
Other languages
German (de)
English (en)
Other versions
EP3065417B1 (fr
Inventor
Tobias Daniel Rosenkranz
Tobias Wurzbacher
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Sivantos Pte Ltd
Original Assignee
Sivantos Pte Ltd
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Sivantos Pte Ltd filed Critical Sivantos Pte Ltd
Publication of EP3065417A1 publication Critical patent/EP3065417A1/fr
Application granted granted Critical
Publication of EP3065417B1 publication Critical patent/EP3065417B1/fr
Active legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Images

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/002—Devices for damping, suppressing, obstructing or conducting sound in acoustic devices
    • 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3026—Feedback
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3028—Filtering, e.g. Kalman filters or special analogue or digital filters
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/02—Circuits for transducers, loudspeakers or microphones for preventing acoustic reaction, i.e. acoustic oscillatory feedback

Definitions

  • the invention relates to a method for suppressing a noise in an acoustic system, wherein the acoustic system comprises at least one microphone and at least one speaker, and wherein the at least one microphone generates an input signal, wherein the at least one loudspeaker generates an acoustic signal, which partially the at least one microphone feeds back.
  • noise caused by feedback may occur.
  • Acoustic feedback may result from the fact that the acoustic signal generated by the speaker is partially perceived by the microphone, and thereby re-entry into the acoustic system.
  • the input signal generated by the microphone is amplified in the acoustic system, so that within the closed loop, which is formed by the speaker, the acoustic signal generated by this, the microphone, and the signal processing within the acoustic system, a signal component by the feedback continues is amplified to a whistling noise when the gain in the signal processing within the acoustic system exceeds a certain limit.
  • k is the discrete time index
  • x is the input to the feedback cancellation system
  • ⁇ is the step size over which the speed of adaptation or convergence is controlled, and * denotes the complex conjugation.
  • the input signal m is often first digitized with a comparatively high sampling rate and thereby converted into time-discrete sampled values. Subsequently, in each case a multiplicity, for example 128, of consecutive samples is combined to form a so-called frame.
  • a spectral analysis of the input signal can now be carried out on the basis of the samples forming the frame by means of Fourier transformation.
  • the window to be considered is shifted in the direction of the time axis by a few sample values, for example 32, so that the windows of the respective samples to be taken into account for one frame overlap partially for adjacent frames.
  • the time index can in this case be understood as a frame index, whereby the adaptive filter can also be used in the frequency domain.
  • the filter coefficients h are vectors whose entries each correspond to a spectral subband.
  • the application is not limited to this case. Details can be found for example in S. Haykin, "Adaptive Filter Theory” (Englewood Cliffs, NJ: Prentice-Hall, 1996 ) or T. v. Waterschoot & M. Moonen, "Fifty years of acoustic feedback control: state of the art and future challenges "(Proc. IEEE, Vol. 99, No. 2, Feb. 2011, pages 288-327 ).
  • correlated input signals such as may be generated by the recording of music or spoken language
  • an adaptive filter which may result in at least partial cancellation of a target signal.
  • This can produce noticeable signal artifacts in the output signal, resulting in a significant deterioration of the sound quality.
  • the whistling noise generated by acoustic feedback also has a high correlation in the signals concerned, particularly when there is a correlated target signal which is picked up and fed back through a loudspeaker after playback. If now an adaptive filter is used to suppress the noise generated thereby, signal components of the target signal can be at least partially extinguished in the suppression of the interference signal of the feedback, which has a negative effect on the sound quality of the output signal.
  • the invention is therefore based on the object of mentioning a method for suppressing an acoustic noise caused by noise, which allows the use of an adaptive filter, and at the same time has the highest possible sound quality in the output signal.
  • the acoustic system comprises at least one microphone and at least one speaker, wherein the at least one microphone generates an input signal and wherein the at least one loudspeaker generates an acoustic signal, which partially feeds back to the at least one microphone, wherein along a main signal path in response to the input signal, a first intermediate signal and from the first intermediate signal by a frequency distortion a Output signal is formed, wherein from the main signal path, the output signal is coupled into a signal feedback path, wherein in the signal feedback path from the output signal by decorrelation, a second intermediate signal is formed, which is used as input for an adaptive filter, which generates a compensation signal, and wherein the compensation signal is supplied to the input signal for compensation, wherein from the input signal and / or from the compensated input signal, a third intermediate signal is formed, which is used as input for the adaptive filter, and wherein the output signal is supplied to the at least one speaker for
  • the output signal can also be used as a further input variable for the adaptive filter, wherein the second intermediate signal and the third intermediate signal are used in the adaptive filter for determining filter coefficients, by means of which the output signal is filtered and the compensation signal is thereby generated.
  • the invention is based on the following considerations: A reduction in the step size ⁇ of an adaptive filter used would result in the filter diverging much more slowly in the case of a correlated input signal, so that unwanted artifacts in the output signal could be reduced or become inaudible.
  • the reduction of the step size could, for example, always take place when a correlated or tonal input signal is registered.
  • a disadvantage of such a procedure is that any change in the acoustic feedback path while registering the correlated signal can not be tracked fast enough to avoid feedback noise, because of the reduced pitch ⁇ constraints on filter adaptability be placed.
  • the step size is therefore always seen as a trade-off between sound quality and the ability to respond to changes in the acoustic feedback path.
  • Another way to solve the problems of an adaptive filter for a strongly correlated input signal is a possible decorrelation of the input signal (so-called "pre-whitening"). Since only correlated input signals cause matching problems in the adaptive filter, such decorrelation could first solve the problem.
  • Such decorrelation is often implemented by a linear predictor. For a correlated input signal, a prediction is made for one or more future samples of the signal as a function of past observed samples of the signal. This prediction is then subtracted from the actual input signal. The result of this subtraction is called the prediction error signal ("residual signal"). For example, a sinusoidal signal is completely deterministic and therefore perfectly predictable. In this case, for a corresponding prediction order, the residual signal would be zero.
  • s (k) represents the sample of the input signal for the prediction at time k
  • a (i) denotes the filter coefficient of decorrelation
  • P the order of prediction.
  • the prediction error signal thus generated is generally of complex value.
  • Interference caused by feedback also has significantly correlated signal components. If a decorrelation is applied to such a signal, the signal strength of the resulting prediction error signal is very low. For reuse in one this would mean that the adaptive filter is not excited at the frequency of the noise generated by the feedback. Thus, the filter can not adapt to the acoustic feedback path at this frequency, whereby the noise remains until the acoustic feedback path changes.
  • the autocorrelation values are time-dependent, and therefore preferred to be determined repeatedly.
  • most non-stationary signals can be considered nearly stationary within a time window of a certain duration. The length of this time window depends on the degree to which the signal is not stationary.
  • the adaptation speed of a filter or estimator which calculates the autocorrelation values of an input signal, plays an important role in this case: the faster the estimator, the better it is not possible to track stationary signals, which improves the decorrelation of an input signal. So a non-stationary signal for decorrelation within a short To be able to treat the time window as stationary requires the fastest possible estimator. This also applies to those decorrelations which use a different method. In the NLMS algorithm, for example, the adaptation speed and thus the ability to decorrelate nonstationary signals is regulated by the step size.
  • a correlated target signal for the adaptive filter for canceling a feedback-related noise is preferably previously decorrelated, but by decorrelating the adaptive filter at the frequencies of the feedback-induced noise is no longer excited, could now be bypassed, that in a first step, such a noise is detected, and in response to such detection in a second step in this case, the decorrelation is omitted.
  • this has several practical disadvantages: First, such a detection in practice is always faulty. In particular, if the acoustical feedback stimulates a plurality of closely spaced frequencies, these may not be sufficiently suppressed due to insufficient spectral resolution in the detection.
  • Another possibility could be to determine the filter coefficients for the decorrelation in another acoustic system, and to transmit these filter coefficients continuously between the participating acoustic systems for adaptation. This possibility would be given in particular in a binaural hearing aid system.
  • the mentioned The idea would be based on the assumption that the sound signals from the environment recorded by the respective acoustic systems have a high degree of similarity, but that noise caused by feedback in a single system affects only the individual acoustic system. Since noise caused by feedback will likely only occur in one acoustic system at a given frequency, the filter coefficients for decorrelation determined in another acoustic system may be considered a good estimate for the decorrelation of a target signal in the feedback acoustic system are used.
  • a time delay of the filter coefficients may occur so that they are no longer up-to-date when received in the respective other acoustic system, or because of the spatial arrangement of the acoustic systems involved, the respective filter coefficients do not constitute a sufficiently good estimate for the respective other system. This can occur, for example, in a binaural hearing aid system due to shading effects caused by the head of the user.
  • the invention proposes to first subject an output signal of the acoustic system, which is to be fed into a signal feedback path, to frequency distortion, and then to decorrelate it.
  • a time-dependent frequency distortion can be used here.
  • Noise caused by feedback usually has a nearly perfectly sinusoidal signal. Due to the frequency distortion, this form is lost. If, for example, a time-dependent frequency shift is selected for the frequency distortion, the signals of the noise follow this frequency shift.
  • the autocorrelation values of frequency-distorted signals decrease with increasing time interval, so that the time window, while the interference caused by feedback can be considered stationary, is shortened.
  • a decorrelator such that it does not adapt to the interference signal of the feedback.
  • the time window in which signals can be considered stationary is preferably to be chosen so that the interference signal of the feedback is not considered stationary by the frequency distortion, the actually non-stationary signal components of a target signal already.
  • the decorrelation is not adapted to the interference signal, but only to the signal components of the target signal, which are decorrelated.
  • the decorrelated signal now removes the non-stationary correlated signal components as they occur in the recording of spoken speech, but not the signal components caused by the feedback.
  • the decorrelated signal is now supplied as an intermediate signal to the adaptive filter, which can generate a compensation signal based on the feedback caused by feedback, which is fed back to the main signal path for the suppression of the noise.
  • the input signal is favorably time-discretized, whereby a "least mean square” algorithm (LMS) is used as the adaptive filter.
  • LMS least mean square algorithm
  • the output signal is preferably used as the reference signal, and the error signal of the LMS filter is formed by the difference between the input signal and the compensation signal.
  • the specified method is particularly advantageous in the use of an LMS algorithm in the adaptive filter, since the frequency distortion of the output signal solves the divergence problems which occur when using an LMS algorithm for the adaptive filtering of feedback-related interference signals.
  • the step size in the LMS algorithm is normalized via the second intermediate signal.
  • This procedure is also called "normalized least mean square" (NLMS).
  • NLMS normalized least mean square
  • the frequency distortion to form the output signal from the first intermediate signal is achieved by a frequency shift.
  • a frequency shift is used.
  • This provides the ability to tune the decorrelator's adaptation speed to the frequency offset, and thus effectively exclude the frequency-shifted signal components of the acoustic feedback noise from the decorrelation.
  • frequency distortion may also be by phase modification, frequency transposition, or non-linear transformation.
  • the adaptation speed of the decorrelator is preferably to be tuned to the respective degree of frequency distortion.
  • the output signal for decoding the second intermediate signal is decorrelated by means of a linear prediction filter.
  • the filter coefficients of the linear prediction filter are preferably to be determined by means of a Levinson-Durbin recursion or by means of an LMS or NLMS algorithm.
  • the advantage of a linear prediction filter is that only linear systems of equations have to be solved, which limits the numerical complexity for the respective filter problem.
  • the input signal or the compensated input signal can also be decorrelated by means of a linear prediction filter, and used to form the third intermediate signal, which is supplied as an input variable to the adaptive filter.
  • time-dependent autocorrelation values of the output signal and / or of an error signal based on the input signal are preferably used for the filter coefficients of the linear prediction filter.
  • the autocorrelation values can be used for a Levinson-Durbin algorithm. Taking into account the time dependence of the autocorrelation values allows the decorrelation to be adjusted to the degree of frequency distortion via the appropriate choice of a corresponding time window, after which the autocorrelation values are again determined.
  • the filter coefficients of, in particular, each linear prediction filter are adapted as a function of the decorrelation strength of the frequency distortion.
  • the time window in which signals can be regarded as stationary depends on the decorrelation strength of the frequency distortion.
  • this can be done, for example, via a repeated adaptation of the autocorrelation values in the mentioned time intervals, from which the filter coefficients are to be determined again.
  • the step size can instead be adjusted accordingly in the time intervals specified.
  • the described functional dependence of the time intervals or of the stationary time window can influence which signal components are still perceived as stationary by the decorrelator, so that the signal components of the interference signal affected by the frequency distortion are not decorrelated.
  • a decorrelator which has too short a "stationary time window", could also perceive signal components of a frequency-distorted originally monofrequency signal as stationary and therefore decorrelate it. This is circumvented by adapting the rate of adaptation of the decorrelation to the degree of frequency distortion, in particular to that of its own decorrelation strength. If, for example, a time-dependent frequency shift is selected, then this is preferably carried out more quickly than signals are considered to be stationary in the time window for the decorrelation.
  • the filter coefficients of the, in particular each linear prediction filter are adapted in dependence on a transfer function of a model of the acoustic system, which comprises the at least one microphone and at least one speaker reproducing the corrected output signal.
  • the time intervals for the adaptation of the filter coefficients may additionally depend on the decorrelation strength of the frequency distortion.
  • the transfer function may include the specific characteristics of the acoustic system, such as gain values in individual sub-bands. In such a model, it is also possible, at least implicitly via coefficients of the transfer function, to enter into the probability that a feedback causes noise at a certain frequency.
  • the decorrelation adaptation rate may be reduced to ensure that the frequency-distorted components of the original monofrequency interference signal are not considered stationary and decorrelated become. If feedback is unlikely, the decorrelator adaptation time window may be shortened so that tonal signal components generated, for example, by voice recording are quickly recognized and decorrelated.
  • the invention further provides an acoustic system comprising at least one microphone for generating an input signal, at least one loudspeaker for reproducing an output signal, and a control unit which is adapted to generate a noise by feedback of the output signal reproduced via the at least one loudspeaker is caused in the input signal generated by the at least one microphone, suppress by the above-described method.
  • the acoustic system is designed as a hearing aid, and advantageously as a hearing aid.
  • FIG. 1 3 shows a schematic block diagram of the sequence of a method 1 for suppressing a noise g in an acoustic system 2.
  • the acoustic system 2 which is given here by a hearing aid 3, for example a hearing aid device comprising a microphone 4 and a loudspeaker 6.
  • the microphone signal m recorded by the microphone 4 is supplied in a main signal path 8 to a signal processing unit 10, where it is amplified inter alia.
  • an output signal xs is output to the microphone 4, which generates an acoustic signal p from the output signal xs.
  • a part of the acoustic signal p generated by the loudspeaker 6 is again recorded by the microphone 4 as feedback fb, and thus finds its way into the microphone signal m.
  • the feedback fb signal components of the acoustic signal p in the microphone signal m are fed again to the signal processing unit 10 and further amplified there.
  • the repeated amplification, playback and recording in a closed process produces noise g in the form of almost monofrequent whistles.
  • the signal feedback path 16 is provided.
  • the output signal x s is coupled out of the main signal path 8 and fed to a decorrelator 18.
  • the decorrelator 18 is formed by a linear prediction filter 20.
  • the signal processing unit 10 outputs a first intermediate signal x, which is converted by a frequency distortion 22 in the output signal xs.
  • the frequency distortion 22, which is achieved in the present case by a frequency shift 23, has the consequence that the linear prediction filter 20 does not decorrelate the signal components corresponding to the noise g, but only signal components of a target signal.
  • a second intermediate signal xw is output as an input to an adaptive filter 24.
  • the adaptive filter 24 generates from the output signal xs a compensation signal c, which is subtracted from the microphone signal m to compensate for the noise g. As a result, the signal feedback path 16 is closed.
  • the adaptive filter 24 is supplied with an additional intermediate signal ew as an input signal.
  • This third intermediate signal ew is formed from the error signal e, which results from the microphone signal m compensated by the compensation signal c.
  • the error signal e is now likewise decorrelated by a linear prediction filter 26, and the decorrelated error signal ew is supplied as a second input variable to the adaptive filter 24.
  • the coefficients h are calculated in a filter block 28 of the adaptive filter 24, from which a signal block 30 of the adaptive filter together with the output signal xs generates the compensation signal c.
  • the frequency shift 23 ensures that the linear prediction filter 20 does not decorrelate any signal components associated with the noise g, whereby the adaptive filter 24 would no longer compensate for these with the compensation signal c.
  • the length of the stationary time window T of the linear prediction filters 20, 26, and thus their adaptation speed, is thereby controlled as a function of the frequency shift 23.
  • a control unit 32 in the hearing aid 3 performs all specified method steps.
  • FIG. 2 is in a block diagram a slight modification of the in FIG. 1 shown method 1 shown.
  • the decorrelated error signal ew which is supplied as an input to the adaptive filter, is formed from an input signal mw decorrelated in the linear prediction filter 26 and a decorrelated compensation signal cw.
  • the decorrelated compensation signal cw is formed in the filter block 28 of the adaptive filter from the error signal ew decorrelated in the linear prediction filter 26 and the second intermediate signal xw, which is given by the output signal xs decorrelated in the linear prediction filter 20.
  • the length of the stationary time window T of the linear prediction filters 20, 26, and thus their adaptation speed is here determined by an adjustment control 34, in which the degree df of the frequency shift 23, the gain n of the signal processing unit 10 in individual sub-bands, and a transfer function, not shown of the acoustic system 2 find input and be used to determine the time window T.
  • an adjustment control 34 in which the degree df of the frequency shift 23, the gain n of the signal processing unit 10 in individual sub-bands, and a transfer function, not shown of the acoustic system 2 find input and be used to determine the time window T.
  • a model of the acoustic feedback path fb determined by the filter coefficients h so that the adaptation speed of the decorrelation to the linear prediction filters 20, 26 is also determined as a function of the feedback estimated by this model.
  • the use of such an adjustment control 34 is not based on the in FIG. 2 illustrated form of the signal feedback path 16 is limited, but can in principle in various embodiments, in particular in. In FIG

Landscapes

  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
EP16151092.0A 2015-03-05 2016-01-13 Procede de suppression d'un bruit parasite dans un systeme acoustique Active EP3065417B1 (fr)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
DE102015204010.0A DE102015204010B4 (de) 2015-03-05 2015-03-05 Verfahren zur Unterdrückung eines Störgeräusches in einem akustischen System

Publications (2)

Publication Number Publication Date
EP3065417A1 true EP3065417A1 (fr) 2016-09-07
EP3065417B1 EP3065417B1 (fr) 2018-11-14

Family

ID=55085608

Family Applications (1)

Application Number Title Priority Date Filing Date
EP16151092.0A Active EP3065417B1 (fr) 2015-03-05 2016-01-13 Procede de suppression d'un bruit parasite dans un systeme acoustique

Country Status (4)

Country Link
US (1) US9824675B2 (fr)
EP (1) EP3065417B1 (fr)
DE (1) DE102015204010B4 (fr)
DK (1) DK3065417T3 (fr)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN106454461A (zh) * 2016-10-21 2017-02-22 安徽协创物联网技术有限公司 一种视频直播回看系统

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE102017203631B3 (de) 2017-03-06 2018-05-17 Sivantos Pte. Ltd. Verfahren zur Frequenzverzerrung eines Audiosignals
US11468873B2 (en) 2017-09-29 2022-10-11 Cirrus Logic, Inc. Gradual reset of filter coefficients in an adaptive noise cancellation system
US11153684B2 (en) * 2018-11-15 2021-10-19 Maxim Integrated Products, Inc. Dynamic debuzzer for speakers

Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2005096670A1 (fr) * 2004-03-03 2005-10-13 Widex A/S Appareil auditif comprenant un systeme adaptatif de suppression de retroaction
EP2086250A1 (fr) * 2008-02-01 2009-08-05 Oticon A/S Système d'écoute avec système d'annulation de rétroaction acoustique, procédé et utilisation
EP2503795A2 (fr) * 2011-03-25 2012-09-26 Siemens Medical Instruments Pte. Ltd. Appareil auditif avec dispositif de suppression de l'effet Larsen et procédé de fonctionnement de l'appareil auditif
DE102013207403B3 (de) * 2013-04-24 2014-03-13 Siemens Medical Instruments Pte. Ltd. Verfahren zur Steuerung einer Adaptionsschrittweite und Hörvorrichtung
EP2736271A1 (fr) * 2012-11-27 2014-05-28 Oticon A/s Procédé de commande d'un algorithme de mise à jour d'un système d'estimation de rétroaction adaptative et unité de dé-corrélation

Family Cites Families (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DK1203509T3 (da) * 1999-07-19 2007-01-02 Oticon As Tilbagekoblingsannullering med anvendelse af båndbreddedetektering
EP2002690B2 (fr) 2006-04-01 2019-11-27 Widex A/S Appareil auditif et procédé permettant de commander la vitesse d'adaptation dans des systèmes anti-rétroaction pour appareils auditifs
EP2148528A1 (fr) * 2008-07-24 2010-01-27 Oticon A/S Filtre de prédiction adaptatif à long terme pour blanchiment adaptatif
KR101225556B1 (ko) 2008-08-25 2013-01-23 돌비 레버러토리즈 라이쎈싱 코오포레이션 프리-화이트닝을 사용한 ｌｍｓ 알고리즘에 의해 적응된 적응형 필터의 갱신된 필터 계수를 결정하기 위한 방법
US8630437B2 (en) * 2010-02-23 2014-01-14 University Of Utah Research Foundation Offending frequency suppression in hearing aids

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2005096670A1 (fr) * 2004-03-03 2005-10-13 Widex A/S Appareil auditif comprenant un systeme adaptatif de suppression de retroaction
EP2086250A1 (fr) * 2008-02-01 2009-08-05 Oticon A/S Système d'écoute avec système d'annulation de rétroaction acoustique, procédé et utilisation
EP2503795A2 (fr) * 2011-03-25 2012-09-26 Siemens Medical Instruments Pte. Ltd. Appareil auditif avec dispositif de suppression de l'effet Larsen et procédé de fonctionnement de l'appareil auditif
EP2736271A1 (fr) * 2012-11-27 2014-05-28 Oticon A/s Procédé de commande d'un algorithme de mise à jour d'un système d'estimation de rétroaction adaptative et unité de dé-corrélation
DE102013207403B3 (de) * 2013-04-24 2014-03-13 Siemens Medical Instruments Pte. Ltd. Verfahren zur Steuerung einer Adaptionsschrittweite und Hörvorrichtung

Non-Patent Citations (4)

* Cited by examiner, † Cited by third party
Title
JOSON H A L ET AL: "ADAPTIVE FEEDBACK CANCELLATION WITH FREQUENCY COMPRESSION FOR HEARING AIDS", THE JOURNAL OF THE ACOUSTICAL SOCIETY OF AMERICA, AMERICAN INSTITUTE OF PHYSICS FOR THE ACOUSTICAL SOCIETY OF AMERICA, NEW YORK, NY, US, vol. 94, no. 6, 31 December 1993 (1993-12-31), pages 3248 - 3254, XP000407303, ISSN: 0001-4966, DOI: 10.1121/1.407231 *
KAWTHER ESSAFI ET AL: "A decorrelation based adaptive prediction filter for acoustic feedback cancellation in hearing aids", INFORMATION SCIENCES SIGNAL PROCESSING AND THEIR APPLICATIONS (ISSPA), 2010 10TH INTERNATIONAL CONFERENCE ON, IEEE, PISCATAWAY, NJ, USA, 10 May 2010 (2010-05-10), pages 69 - 72, XP031777918, ISBN: 978-1-4244-7165-2 *
S. HAYKIN: "Adaptive Filter Theory", 1996, PRENTICE-HALL
T. V. WATERSCHOOT; M. MOONEN: "Fifty years of acoustic feedback control: state of the art and future challenges", PROC. IEEE, vol. 99, no. 2, February 2011 (2011-02-01), pages 288 - 327, XP055225971, DOI: doi:10.1109/JPROC.2010.2090998

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN106454461A (zh) * 2016-10-21 2017-02-22 安徽协创物联网技术有限公司 一种视频直播回看系统

Also Published As

Publication number Publication date
US9824675B2 (en) 2017-11-21
US20160260423A1 (en) 2016-09-08
EP3065417B1 (fr) 2018-11-14
DK3065417T3 (da) 2019-03-04
DE102015204010A1 (de) 2016-09-08
DE102015204010B4 (de) 2016-12-15

Similar Documents

Publication Publication Date Title
DE69738288T2 (de) Einrichtung zur unterdrückung einer störenden komponente eines eingangssignals
DE112009000805B4 (de) Rauschreduktion
DE60310725T2 (de) Verfahren und vorrichtung zur verarbeitung von subbandsignalen mittels adaptiver filter
DE112017006486T5 (de) Online-enthallungsalgorithmus basierend auf gewichtetem vorhersagefehler für lärmbehaftete zeitvariante umgebungen
EP2158588B1 (fr) Procédé de filtrage spectral de signaux parasités
EP1386307B2 (fr) Procede et dispositif pour determiner un niveau de qualite d'un signal audio
EP1143416A2 (fr) Suppression de bruit dans le domaine temporel
EP1091349A2 (fr) Procédé et dispositif pour la réduction de bruit durant la transmission de parole
DE112017007005B4 (de) Akustiksignal-verarbeitungsvorrichtung, akustiksignalverarbeitungsverfahren und freisprech-kommunikationsvorrichtung
EP3065417B1 (fr) Procede de suppression d'un bruit parasite dans un systeme acoustique
DE102008042579A1 (de) Verfahren zur Fehlerverdeckung bei fehlerhafter Übertragung von Sprachdaten
EP3089481B1 (fr) Procédé de suppression du bruit d'un signal d'entrée en fonction de la fréquence
EP3197181B1 (fr) Procédé de réduction du temps de latence d'un banc de filtrage destiné au filtrage d'un signal audio et procédé de fonctionnement sans latence d'un système auditif
DE102015216822B4 (de) Verfahren zur Unterdrückung einer Rückkopplung in einem Hörgerät
EP1189419A2 (fr) Procede et appareil pour eliminer l'interference d'un haut-parleur sur de signaux de microphone
EP2080197B1 (fr) Dispositif d'élimination du bruit dans un signal audio
DE102008017550A1 (de) Mehrstufiges Schätzverfahren zur Störgeräuschreduktion und Hörvorrichtung
DE69817461T2 (de) Verfahren und Vorrichtung zur optimierten Verarbeitung eines Störsignals während einer Tonaufnahme
DE60033039T2 (de) Vorrichtung und verfahren zur unterdrückung von zischlauten unter verwendung von adaptiven filteralgorithmen
DE112021001228T5 (de) Signalverarbeitungsvorrichtung, signalverarbeitungsverfahren und programm
DE102018117558A1 (de) Adaptives nachfiltern
DE102019105458B4 (de) System und Verfahren zur Zeitverzögerungsschätzung
DE3230391C2 (fr)
DE4445983C2 (de) Verfahren zur Rauschunterdrückung und Vorrichtungen zur Durchführung der Verfahren
DE102018117556A1 (de) Einzelkanal-rauschreduzierung

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): AL AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO RS SE SI SK SM TR

AX Request for extension of the european patent

Extension state: BA ME

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: REQUEST FOR EXAMINATION WAS MADE

17P Request for examination filed

Effective date: 20170109

RBV Designated contracting states (corrected)

Designated state(s): AL AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO RS SE SI SK SM TR

RAP1 Party data changed (applicant data changed or rights of an application transferred)

Owner name: SIVANTOS PTE. LTD.

GRAP Despatch of communication of intention to grant a patent

Free format text: ORIGINAL CODE: EPIDOSNIGR1

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: GRANT OF PATENT IS INTENDED

RIC1 Information provided on ipc code assigned before grant

Ipc: G10K 11/00 20060101ALN20180702BHEP

Ipc: H04R 25/00 20060101ALI20180702BHEP

Ipc: H04R 3/02 20060101ALI20180702BHEP

Ipc: H04R 1/10 20060101AFI20180702BHEP

INTG Intention to grant announced

Effective date: 20180719

GRAS Grant fee paid

Free format text: ORIGINAL CODE: EPIDOSNIGR3

GRAA (expected) grant

Free format text: ORIGINAL CODE: 0009210

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: THE PATENT HAS BEEN GRANTED

AK Designated contracting states

Kind code of ref document: B1

Designated state(s): AL AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HR HU IE IS IT LI LT LU LV MC MK MT NL NO PL PT RO RS SE SI SK SM TR

REG Reference to a national code

Ref country code: CH

Ref legal event code: EP

Ref country code: AT

Ref legal event code: REF

Ref document number: 1066262

Country of ref document: AT

Kind code of ref document: T

Effective date: 20181115

REG Reference to a national code

Ref country code: DE

Ref legal event code: R096

Ref document number: 502016002461

Country of ref document: DE

REG Reference to a national code

Ref country code: IE

Ref legal event code: FG4D

Free format text: LANGUAGE OF EP DOCUMENT: GERMAN

REG Reference to a national code

Ref country code: CH

Ref legal event code: NV

Representative=s name: E. BLUM AND CO. AG PATENT- UND MARKENANWAELTE , CH

REG Reference to a national code

Ref country code: DK

Ref legal event code: T3

Effective date: 20190225

REG Reference to a national code

Ref country code: NL

Ref legal event code: MP

Effective date: 20181114

REG Reference to a national code

Ref country code: LT

Ref legal event code: MG4D

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: BG

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20190214

Ref country code: IS

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20190314

Ref country code: LT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: FI

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: LV

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: HR

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: ES

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: NO

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20190214

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: PT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20190314

Ref country code: GR

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20190215

Ref country code: NL

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: SE

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: RS

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: AL

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: CZ

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: IT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: PL

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

REG Reference to a national code

Ref country code: DE

Ref legal event code: R097

Ref document number: 502016002461

Country of ref document: DE

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: MC

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: SK

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: RO

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: EE

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

Ref country code: SM

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

PLBE No opposition filed within time limit

Free format text: ORIGINAL CODE: 0009261

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: NO OPPOSITION FILED WITHIN TIME LIMIT

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: LU

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20190113

REG Reference to a national code

Ref country code: BE

Ref legal event code: MM

Effective date: 20190131

26N No opposition filed

Effective date: 20190815

REG Reference to a national code

Ref country code: IE

Ref legal event code: MM4A

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: SI

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: BE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20190131

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: IE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20190113

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: TR

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: MT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: CY

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: HU

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT; INVALID AB INITIO

Effective date: 20160113

REG Reference to a national code

Ref country code: AT

Ref legal event code: MM01

Ref document number: 1066262

Country of ref document: AT

Kind code of ref document: T

Effective date: 20210113

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: AT

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20210113

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: MK

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20181114

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: FR

Payment date: 20230123

Year of fee payment: 8

Ref country code: DK

Payment date: 20230123

Year of fee payment: 8

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: DE

Payment date: 20240119

Year of fee payment: 9

Ref country code: GB

Payment date: 20240124

Year of fee payment: 9

Ref country code: CH

Payment date: 20240202

Year of fee payment: 9