EP2981099A2 - Procede et dispositif de suppression de l'effet larsen - Google Patents

Procede et dispositif de suppression de l'effet larsen Download PDF

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Publication number
EP2981099A2
EP2981099A2 EP15178938.5A EP15178938A EP2981099A2 EP 2981099 A2 EP2981099 A2 EP 2981099A2 EP 15178938 A EP15178938 A EP 15178938A EP 2981099 A2 EP2981099 A2 EP 2981099A2
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EP
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Prior art keywords
feedback
function
transfer function
adaptive filter
hearing aid
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EP15178938.5A
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German (de)
English (en)
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EP2981099A3 (fr
EP2981099B1 (fr
Inventor
Tobias Daniel Rosenkranz
Henning Puder
Tobias Wurzbacher
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Sivantos Pte Ltd
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Sivantos Pte Ltd
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Electric hearing aids
    • 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; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/41—Detection or adaptation of hearing aid parameters or programs to listening situation, e.g. pub, forest
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2460/00—Details of hearing devices, i.e. of ear- or headphones covered by H04R1/10 or H04R5/033 but not provided for in any of their subgroups, or of hearing aids covered by H04R25/00 but not provided for in any of its subgroups
    • H04R2460/01—Hearing devices using active noise cancellation

Definitions

  • the invention relates to a method for feedback suppression and to a device for carrying out the method.
  • a feedback transfer function is estimated, coefficients of an adaptive filter for suppressing feedback are adjusted, and the adaptive filter is applied to a signal derived from an acousto-electrical transducer input signal.
  • Hearing aids are portable hearing aids that are used to care for the hearing impaired.
  • different types of hearing aids such as behind-the-ear hearing aids (BTE), hearing aid with external receiver (RIC: receiver in the canal) and in-the-ear hearing aids (ITE), e.g. Concha hearing aids or canal hearing aids (ITE, CIC).
  • BTE behind-the-ear hearing aids
  • RIC hearing aid with external receiver
  • ITE in-the-ear hearing aids
  • ITE in-the-ear hearing aids
  • ITE in-the-ear hearing aids
  • ITE concha hearing aids or canal hearing aids
  • the hearing aids listed by way of example are worn on the outer ear or in the ear canal.
  • bone conduction hearing aids, implantable or vibrotactile hearing aids are also available on the market. The stimulation of the damaged hearing takes place either mechanically or electrically.
  • Hearing aids have in principle as essential components an input transducer, an amplifier and an output transducer.
  • the input transducer is usually an acoustoelectric transducer, z. As a microphone, and / or an electromagnetic receiver, for. B. an induction coil.
  • the output transducer is usually used as an electroacoustic transducer, z. As miniature speaker, or as an electromechanical transducer, z. B. bone conduction, realized.
  • the amplifier is usually integrated in a signal processing device.
  • the power is usually supplied by a battery or a rechargeable battery.
  • the respective change of coefficients of the adaptive filter is determined by means of a mathematical method after a normalized least deviation of the square error (NMLS).
  • NMLS normalized least deviation of the square error
  • the speed with which the adaptive filter can adapt is influenced by a step size ⁇ . If the step size is large, the adaptive filter can follow quickly, if the step size is small, the filter better maps the input function with slight changes.
  • the object of the present invention is therefore to provide a method and an apparatus in which a feedback suppression is improved.
  • the inventive method relates to a method for reducing feedback in a hearing aid.
  • the hearing aid has an acousto-electrical converter, a signal processing device, a feedback suppression device and an electro-acoustic transducer.
  • a first feedback transfer function is determined at a first time.
  • the feedback transfer function forms feedback paths from the signal processing device via the electro-acoustic transducer, an acoustic signal path from the electro-acoustic transducer to the acousto-electrical transducer and via the acousto-electrical transducer back to the signal processing device.
  • the acoustic signal path depends on the environment of the head and changes, for example, as the wearer moves.
  • the determination can include, for example, measuring different feedback transfer functions in a laboratory or estimating by means of approximation methods such as NLMS in the operation of the hearing aid aid at the wearer's ear.
  • a weighted average function and / or a plurality of impulse response parameters is determined as a function of amplitude amounts of the first feedback transfer function.
  • an enveloping function for the magnitude of the amplitudes may be formed or a function of the amplitude squares smoothed by low pass or bandpass, which reflects an energy of the impulse response over a time delay with respect to the impulse excitation.
  • the impulse response parameters are resolved over a time delay with respect to the impulse excitation, i.e. at different values for the time delay different impulse response parameters are determined.
  • individual impulse response parameters are preferably determined as a function of different function values of the enveloping function for the magnitude of the amplitudes or the function of the amplitude squares which is smoothed by means of lowpass or bandpass.
  • the impulse response parameters are determined from the weighted average function, which depends on the first feedback transfer function.
  • the weighted average function in this case forms a weighted average over the first feedback transfer function and other feedback transfer functions, with the averaging preferably occurring pointwise to the individual time delays after which the feedback functions are resolved.
  • the impulse response parameters preferably have an immediate dependence on the impulse response of a feedback path which is mapped by the first feedback transfer function or by a weighted average function of a plurality of feedback transfer functions.
  • the impulse response of a feedback path is in this case in particular given by a time-resolved amplitude which has a signal excited in the feedback path by a test pulse.
  • a second feedback transfer function is estimated by means of an adaptive filter.
  • the estimation is done at a second, different time.
  • coefficients of the adaptive filter for suppressing a feedback signal as a function of the weighted average function are determined and / or updated as a function of the impulse response parameters, wherein an adaptation speed of the adaptive filter is formed by a function of the impulse response parameters.
  • a current estimator is made up of estimates of the past and an estimate of the deviation of the estimates of the past from the real values.
  • an impulse response it is possible, for example, to consider respective components with different delay in different coefficients.
  • the weighting of the change in the different coefficients may be weighted as a function of empirical values resulting from mean value functions of exemplary or past impulse responses.
  • the adaptation speed of the adaptive filter is the speed with which the adaptive filter responds to changes in the feedback transmission function to be estimated and thus "adapts" it to the changes.
  • the adaptive filter responds quickly to changes in the feedback function to be imaged Feedback path, whereby suggestions caused by the changes can be quickly suppressed.
  • the adaptive filter is more stable, so that audible artifacts due to the higher inertia in an output signal can be better avoided by the feedback suppression.
  • the function of the impulse response parameters for the adaptation speed is such that for time delays with respect to impulse excitation in which there is a comparatively strong impulse response of the feedback path underlying the impulse response parameters, the adaptive filter quickly adapts to changes in the feedback path, while the adaptive Filter with time delays with respect to a pulse excitations, in which there is no significant impulse response of a feedback path underlying the impulse response parameters, slower adapted to changes in the feedback path.
  • This is achieved, for example, by using as the impulse response parameter a monotonic function of the temporally smoothed amplitude magnitudes of the impulse response in the underlying feedback path, and the adaptation speeds at different time delays with respect to impulse excitation are each formed by the same monotonic function of the corresponding impulse response parameter.
  • the adaptive filter which estimates the second feedback transfer function by its coefficients, makes changes to the estimated feedback path particularly fast, especially where it has a high impulse response.
  • the impulse response parameters are not from the second feedback transfer function itself but from the first feedback transfer function or a weighted average function, which is preferably to be selected as a typical representative of a feedback transfer function possible in the given listening situation with a corresponding feedback path, misadaptation, for example due to tonal excitations in the feedback path, can be avoided since updating the coefficients is no longer limited to the erroneous estimation, but now also depends on an external reference.
  • the adaptive filter is applied to a signal derived from an acoustic input signal of the acousto-electrical transducer.
  • a signal derived from an acoustic input signal of the acousto-electrical transducer For example, it is conceivable to filter out or suppress a feedback component from the acoustic signal by means of the adaptive filter, in that the adaptive filter admits the audio signal with an approximately identical signal with an inverse sign to the feedback component.
  • the method of the present invention advantageously allows for a faster and more accurate estimation of the current feedback transfer function and thus more effective and accurate suppression of feedbacks while reducing Artifacts due to feedback suppression.
  • the coefficients of the adaptive filter are adjusted to ensure rapid adaptation in those regions of the feedback impulse response which involve a high amount of energy, whereas low energy regions undergo only slow adaptation. Low energy areas do not contribute to the risk of feedback whistling, so it is important in these areas to provide for a high degree of artifact freedom through slow adaptation.
  • an enveloping function it is ensured Regions near zero-crossings in the feedback impulse response do not erroneously result in slow adaptation. Temporal averaging ensures that short-term fluctuations do not lead to erroneous adaptations.
  • the hearing aid according to the invention for carrying out the method shares the advantages of the method according to the invention.
  • a multiplicity of feedback transmission functions are determined at different times and the weighted average value function is determined as a function of the multiplicity of the feedback transmission functions.
  • the feedback suppression device forms an average value function from feedback transfer functions over a relatively long period of time, or, in particular, takes account of feedback transfer functions with greatly different properties.
  • the determination of the first and second feedback transfer functions is performed by estimating the feedback transfer functions in the hearing aid.
  • the hearing aid may thus adapt to the wearer's environment during operation and provide better functionality with less feedback and artifacts.
  • the determination of the first feedback transfer function is performed by measuring the feedback transfer functions.
  • the feedback suppression device is implemented as part of the signal processing device, so that the signal processing device carries out the steps of the method.
  • the number of components of the hearing aid can thus be reduced and synergies used in determining the coefficients, for example by accessing common data.
  • this is carried out in a plurality of disjoint or partially overlapping frequency ranges.
  • the step of determining a weighted average function is continued, wherein the second feedback transfer function is used together with the first feedback transfer function to form the weighted average function and a new second feedback transfer function is estimated.
  • the impulse response parameters are determined by a smoothing function of the amplitude amounts as a function of the first feedback transfer function.
  • the first feedback transfer function and a weighted average value function of different feedback transfer functions are included.
  • the feedback transfer function or the weighted average value function is embodied as an impulse response function, so that a smoothing function of the amplitude amounts represents a preferably temporal smoothing of the magnitudes of the impulse responses of the feedback path corresponding to the feedback transfer function at different time delays with respect to the impulse excitation.
  • the smoothing function is preferably designed as an envelope of the amplitude amounts.
  • the envelope is normalized with respect to a reference value dependent on the adaptive filter or with respect to a maximum value for the amplitude amounts.
  • a temporal smoothing of a function of the amplitude amounts which is based on the impulse response parameters, it can be achieved that an impulse response parameter is not influenced by a zero crossing of an oscillating amplitude with strong absolute values in the corresponding range falling randomly on the corresponding time delay, and thus An adaptation speed would not be mistakenly chosen too low for the corresponding time delay.
  • the adaptation speed of the adaptive filter in this range is reduced.
  • the first feedback transfer function or the weighted average function, which underlies the impulse response parameters, preferably represents a typical representative of a feedback transfer function possible in the given listening situation with a corresponding feedback path.
  • the amplitude amounts monotone This means that such a feedback path usually provides correspondingly decreasing contributions to the feedback in this area. Accordingly, the adaptation speed in the estimation of the second feedback transfer function for these areas is also reduced.
  • the coefficients of the adaptive filter are updated by means of an NLMS algorithm, wherein the entries of a vector-valued step size of the NLMS algorithm for updating the coefficients of the adaptive filter are formed on the basis of the impulse response parameters, and wherein the impulse response parameters are based on a smoothing function of the amplitude amounts are determined in dependence on the first feedback transfer function.
  • a Normalized Least Mean Squares (NLMS) algorithm is a filter that is used particularly frequently for suppressing feedback, which updates existing coefficients of the filter as a function of an output signal and an error signal over a step size.
  • the individual coefficients of the filter are hereafter with their corresponding timing - ie the time delay with respect to a pulse excitation - applied to a signal derived from the input signal.
  • the device according to the invention shares the advantages of the method according to the invention.
  • Fig. 1 shows a hearing aid 100 according to the invention as a schematic representation in function blocks.
  • the hearing aid according to the invention has an acousto-electrical converter 2, which converts a mechanical vibration, usually recorded as airborne sound d (k), into an electrical signal m (k).
  • the acousto-electrical converter 2 is one or more microphones, mostly of a micromechanical design, usually capacitive and partly also made of silicon as a MEMS microphone. It is conceivable that the signals of several microphones are connected together as a microphone with directional characteristics.
  • the signal m (k) is preferably a signal having a directional characteristic.
  • the hearing aid 100 further comprises a signal processing device 3, which is designed to amplify an incoming signal e (k) preferably frequency-dependent, so that a hearing impairment of a wearer can be compensated and quiet sounds are raised below the hearing threshold of the wearer in a range above the hearing threshold ,
  • the signal processing device 3 may have, for example, a filter bank.
  • Conceivable further functions of the signal processing device 3 are dynamic compression, classification of listening situations, noise suppression, control of directional characteristics of the microphone, binaural signal processing when the hearing aid 100 is in signal communication with a second hearing aid 100 via a communication interface, not shown.
  • the hearing aid has an electro-acoustic transducer 4, which is designed as a speaker or listener.
  • the electro-acoustic transducer 4 may be arranged in a housing behind the ear in a behind-the-ear hearing aid device 100 and the sound transmitted via a sound tube to an ear piece in the ear canal of the wearer. It is also conceivable in the case of a BTE hearing aid that the electro-acoustic transducer 4 is arranged in the auditory canal of the wearer and receives a signal to be output via an electrical signal connection.
  • the hearing aid device 100 can also be an in-the-ear or CiC (complete in channel) hearing aid device, so that all components of the hearing aid device are arranged on or in the auditory canal of the wearer.
  • the feedback path may be formed by the air, for example by a gap between the auditory canal and a seal of the ear canal (eg an ear cup or an "ear dome") or as a structure-borne sound transmission through a housing of the hearing aid 100. Also conceivable is a combination of both Ways.
  • the properties of the feedback path are also dependent on the environment of a head of the wearer, for example, from a reflection on a wall or a car window or a telephone receiver near the ear.
  • the hearing aid device 100 has a feedback suppression device 6, which in the illustrated Embodiment has an adaptive filter 7 and a mixer 8.
  • the adaptive filter 7 receives the input signal e (k) supplied to the signal processing device 3 via a first signal line 11 and the signal x (k) output by the signal processing device via a second signal line 9. Furthermore, the adaptive filter 7 is connected to the signal processing device 3 via a third signal line 10 in order to detect its effect for processing the input signal e (k). This can be done for example by a transmission of processing parameters.
  • the adaptive filter 7 processes the supplied signals into a compensation signal c (k) which is mixed via a mixer 8 to the electrical signal m (k) in order to reduce feedback. Further details of the manner of generating the compensation signal c (k) is given below Fig.2 explained in more detail.
  • the allocation of functionalities in the Fig. 1 is only an example. It is also conceivable that the feedback suppression unit 6 not as in Fig. 1 shown as a separate function blocks 7 and 8 is executed, but only as program-controlled functions in the signal processing device 3, or as hardware-implemented circuits therein. It is also conceivable that the adaptive filter 7 does not filter by generating a compensation signal c (k) and mixing it with the electrical signal m (k) to reduce a feedback signal by destructive interference, but as a subtractive filter even in the signal path m (k) is provided. Also, the signals x (k) and e (k) can be taken at different points of the signal flow, without departing from the principle of the invention.
  • the adaptive filter 3 determines the influence of the signal processing device 3 by comparing the signals e (k) and x (k) itself.
  • the adaptive filter 7 transmits all information about the function of the signal processing 3 receives the signal connection 10, but only one of the signals e (k) or x (k).
  • Fig. 2 shows an exemplary sequence of a method according to the invention on a hearing aid of the Fig. 1 ,
  • a first feedback transfer function at a first time on a feedback path from the signal processing device 3 via the electro-acoustic transducer 4, an acoustic signal path g (k) from the electro-acoustic transducer 4 to the acousto-electric converter 2 and detects the acousto-electrical converter 2 back to the signal processing device 3.
  • the feedback transfer function be measured in a hearing aid in a measuring box or in a laboratory by measurement on the carrier or a dummy head.
  • the feedback transfer function can be measured more accurately because each input and output signals can be externally detected and processed with each other. It is conceivable to represent typical listening environments, such as phoning with a mobile phone or sitting in a car with the ear near a disc.
  • multiple feedback transfer functions are measured for typical environments.
  • the feedback transmission functions in the hearing aid itself are estimated when worn, i. are detected by step S30 or S30 'explained approximate functions.
  • the thus detected feedback transfer functions advantageously have no influence of the measurement environment and can correspond to everyday situations of the wearer.
  • Fig. 3 illustrates two exemplary impulse responses as a possible representation of a feedback transfer function.
  • impulse response and feedback transfer function in the sense equivalent to each other, that in each case one can be clearly derived from the other by means of mathematical methods.
  • the time is given in multiples of a sampling cycle, in the y-axis a normalized amplitude.
  • the x-axis indicates a time delay with respect to an excitation pulse.
  • step S20 from the detected first feedback transfer functions, a weighted average function is determined in response to amplitude amounts of the first feedback transfer function.
  • a step S20 ' a plurality of impulse response parameters are determined in response to amplitude amounts of the first feedback transfer function. If step S20 'is performed alternatively to step S20, the impulse response parameters are determined directly from the feedback transfer function detected in step S10. If step S20 'is immediately following step S20, the impulse response parameters are determined from a weighted average function of a plurality of feedback transfer functions including the first feedback transfer function detected in step S10.
  • Fig. 4 shows first for each impulse response a function which is generated by normalizing a function in dependence on the amplitude amounts. The functions therefore only have a positive sign. For the large amplitudes at the beginning, the function value is set equal to 1 in the sense of a limitation.
  • An average can be made, for example, by smoothing the feedback transfer function by forming an envelope of the positive amplitudes. Also conceivable is a low-pass or bandpass over a function of the amplitude squares.
  • An average may additionally be added in terms of arithmetic averaging or other averaging, for example by adding of a plurality of function values of different feedback transfer functions and parts are formed by the number of the detected functions, if a plurality of feedback transfer functions have been detected. This can be done for example by measurement or by an iteration of the method via a plurality of feedback transfer functions.
  • other forms are also conceivable, such as the weighting of a function in averaging as a function of the age of the corresponding feedback transfer function.
  • the mean value function can already be calculated outside the hearing aid device 100 in a measuring device and transmitted to the hearing aid device 100. If, on the other hand, it is a feedback transfer function estimated in the hearing aid 100, the weighted average function is preferably used in the hearing aid 100, e.g. determined by the feedback suppression device 6.
  • a second feedback transfer function is estimated.
  • the adaptive filter 7 models the time-dependent feedback transfer function as a time-dependent impulse response g (k) of the feedback path.
  • K indicates a discrete time scale
  • x is the input value of the feedback suppression device
  • ⁇ is a step that controls a matching speed of the filter
  • * denotes the complex conjugate of a value.
  • the coefficients of the adaptive filter for suppressing a feedback signal are adapted to the second feedback transfer function or, in other words, the feedback transfer function is modeled by the coefficients, weighting a change in the coefficients depending on the mean value function and the impulse response parameters, respectively.
  • a correction value is weighted with a weighting factor or a step size. In the illustrated embodiment, this weighting occurs over the pitch ⁇ which, as illustrated above, is used in the estimation of the feedback transfer function modeled by the coefficients.
  • the weighting factor is derived from the mean value function via the impulse response parameters. In the simplest case it could be the value of an in Fig. 4 be shown mean value function itself.
  • the value of a weighting factor ⁇ (k) is then, for example, a function value of an in Fig. 4 shown function for the value k in the x-axis.
  • Fig. 5 For this purpose, instead of a linear, normalized scale to 1, a scale is plotted in accordance with the log 10 log 10 . In this way, the dynamic range of the step size is much larger, so that for large values of the impulse response in Fig. 3 a fast convergence is achieved, while at small values a high accuracy in the adaptation and thus low artifacts occur.
  • the estimation of the second feedback transfer function takes place separately from a weighting of the coefficients in succession.
  • the adaptive filter is applied to a signal derived from an acoustic input signal of the acousto-electrical converter.
  • a hearing aid signal processing such as A / D conversion, amplification, frequency dependent, forming a directivity or other functions that are possible in the signal processing 3.
  • the application of the filter is represented by the compensation signal c (k), which represents an estimated feedback signal and is added with opposite sign to the microphone's signal m (k), so that ideally the signal of the adapted filter and the feedback component of the microphone signal m ( k) cancel.
  • step S40 this is continued after step S40 with step S20, wherein the second feedback transfer function is used in common with the first feedback transfer function to form the average value function and a new second feedback transfer function is estimated in step S30.
  • the steps S10 to S40 are each carried out in separate or only partially overlapping frequency bands, so that different feedback conditions in different frequencies can each be optimally suppressed.
  • a filter bank may be provided in the feedback suppression device 6, or else a filter bank may be used in the signal processing device 3.
  • excitation in the form of a tonal input signal may result in mis-adaptation.
  • the adaptive filter provides as a solution the feedback transfer function of the respective feedback path to which an error term is added, which depends on the autocorrelation of the input signal. Due to the comparatively high autocorrelation of a tonal input signal, in this case an incorrect adaptation to the excitation in the form of the tonal input signal can usually not be sufficiently suppressed by conventional means.
  • FIG Fig. 6 The behavior or responsiveness to changes in the feedback path allowed by the method is illustrated by two diagrams in FIG Fig. 6 shown.
  • the diagrams each show the system distance, which is defined as
  • the system distance is hereby a measure of how far the coefficients h (k) of the adaptive filter correspond to the actual impulse response g (k) in the feedback path.
  • a good match is characterized by values near zero for the system distance.
  • the excitation underlying the feedback path is white noise.
  • a uniform step size ⁇ was used in updating the coefficients h (k) of the adaptive filter.
  • the step size ⁇ was adjusted in the described manner across the individual coefficients to the impulse response of a typical feedback path.
  • FIG Fig. 7 The improvement of the stability of the feedback suppression, in particular the reduction of false adaptations, by the updating of the coefficients h (k) of the adaptive filter by means of individual step widths is illustrated by the diagram in FIG Fig. 7 clear:
  • the system instance is plotted against a time axis scaled in seconds, the three illustrated scenarios being given by: the classical NLMS algorithm and an update of the coefficients with constant increment (upper line 18), an updating of the coefficients by individual, but not time-dependent step sizes (middle line 19), and updating the coefficients by individual, time-dependent step sizes as a function of a weighted averaging "learned" feedback path (lower line 20).

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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)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Filters That Use Time-Delay Elements (AREA)
  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
EP15178938.5A 2014-08-01 2015-07-29 Procede et dispositif de suppression de l'effet larsen Active EP2981099B1 (fr)

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DE102014215165.1A DE102014215165A1 (de) 2014-08-01 2014-08-01 Verfahren und Vorrichtung zur Rückkopplungsunterdrückung

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EP2981099A3 EP2981099A3 (fr) 2016-03-16
EP2981099B1 EP2981099B1 (fr) 2022-12-28

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EP (1) EP2981099B1 (fr)
CN (1) CN105323692B (fr)
AU (1) AU2015207943A1 (fr)
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DK (1) DK2981099T3 (fr)

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C. ANTWEILER; A. SCHIFFER; M. DÖRBECKER: "Accoustic Echo Control with Variable Individual Step Size", PROC. IWAENC, 1995, pages 15 - 18
S. HAYKIN: "Englewood Cliffs", 1996, PRENTICE-HALL, article "Adaptive Filter Theory"
TOON VAN WATERSCHOOT; MARC 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 (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP4132009A3 (fr) * 2021-08-05 2023-02-22 Oticon A/s Dispositif d'aide auditive comprenant un système de commande de rétroaction
US12363487B2 (en) 2021-08-05 2025-07-15 Oticon A/S Hearing device comprising a feedback control system

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US20180041846A1 (en) 2018-02-08
CN105323692A (zh) 2016-02-10
CN105323692B (zh) 2019-02-22
AU2015207943A1 (en) 2016-02-18
US9872114B2 (en) 2018-01-16
US10334371B2 (en) 2019-06-25
DE102014215165A1 (de) 2016-02-18
US20160037269A1 (en) 2016-02-04
EP2981099A3 (fr) 2016-03-16
EP2981099B1 (fr) 2022-12-28

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