EP3340656B1 - Procédé de fonctionnement d'un dispositif de correction auditive - Google Patents

Procédé de fonctionnement d'un dispositif de correction auditive Download PDF

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Publication number
EP3340656B1
EP3340656B1 EP17207540.0A EP17207540A EP3340656B1 EP 3340656 B1 EP3340656 B1 EP 3340656B1 EP 17207540 A EP17207540 A EP 17207540A EP 3340656 B1 EP3340656 B1 EP 3340656B1
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Prior art keywords
signal
frequency
parameter
output
input
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German (de)
English (en)
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EP3340656A1 (fr
Inventor
Tobias Daniel Rosenkranz
Sebastian BEST
Tobias Wurzbacher
Christoph LÜKEN
Stefan Petrausch
Nicola CEBULLA
Christos OREINOS
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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
    • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
    • 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
    • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
    • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
    • 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
    • H04R25/35—Electric hearing aids using translation techniques
    • H04R25/353—Frequency, e.g. frequency shift or compression
    • 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
    • H04R25/43—Electronic input selection or mixing based on input signal analysis, e.g. mixing or selection between microphone and telecoil or between microphones with different directivity characteristics
    • 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
    • H04R25/00—Electric hearing aids
    • H04R25/60—Mounting or interconnection of hearing aid parts, e.g. inside tips, housings or to ossicles
    • 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

Definitions

  • the invention relates to a method for operating a hearing aid, which comprises at least one input converter and at least one output converter, an input signal being generated from a sound signal from the environment by the at least one input converter, and a first intermediate signal being generated as a function of the input signal by means of signal processing ,
  • a sound signal from the environment is typically converted into an electrical signal by means of an input converter, and is processed in a signal processing unit in accordance with the audiological requirements of the user and, in particular, is amplified as a function of frequency.
  • the processed signal is then converted by an output converter into an output sound signal which is fed to the hearing of the user.
  • the situation can arise during operation, even when the hearing aid is used as intended, that the output sound signal of the hearing aid is superimposed by the sound signal of the surroundings when it hits the hearing of the user.
  • a small hole can optionally also be provided in the housing of the hearing device.
  • the input signal generated by the input converter from the sound signal in the environment is now experienced in the signal processing unit, in particular in processes for frequency band filtering, a time delay which cannot be reduced arbitrarily by technical measures of signal processing.
  • EP 2 590 437 A1 discloses a method for the adaptive suppression of acoustic feedback in a hearing aid, the adaptation process being activated periodically, so that in the active state an adaptive filter with a variable step size maps the acoustic feedback path.
  • An algorithm for frequency shift or frequency compression can be started.
  • the duration of an activity or inactivity state can also be changed depending on a listening situation.
  • the DE 10 2010 025 918 A1 mentions a method in which when an acoustic feedback is detected in a hearing aid to better suppress it, a frequency shift is applied to the output signal to be output by the loudspeaker.
  • the EP 2 369 859 A2 mentions a method in a hearing aid in which parameters for a frequency modification of a signal in the hearing aid are changed as a function of hearing environments, and the frequency modification is reduced or completely switched off, for example in a noisy environment, when making calls or when the user speaks. In this way, disruptive artefacts between a frequency-modified output signal and the direct sound that strikes the user's hearing can be reduced or avoided.
  • the invention is therefore based on the object of specifying a method for operating a hearing aid by means of which the unpleasant consequences of comb filter effects for the user can be avoided in the simplest possible way without substantially changing or even impairing the user-specific signal processing.
  • a method for operating a hearing device which comprises at least one input transducer and at least one output transducer, an input signal being generated by the at least one input transducer from a sound signal from the environment, with the input signal being used to classify a hearing situation of the environment takes place and / or for the sound signal of the environment at least one of the four parameters tonality, volume, stationarity and reverberation time are determined, a first intermediate signal being derived as a function of the input signal by means of signal processing, with the classification of the listening situation or with at least one the four parameters tonality, Volume, stationarity and reverberation time, at least one parameter of frequency distortion is specified, and the frequency distortion specified in this way is applied to the first intermediate signal, a comb filter being used based on the classification of the listening situation or on the basis of the at least one of the four parameters tonality, volume, stationarity and reverberation time -Parameter is determined, which indicates a probability value for
  • An input transducer generally includes an acousto-electrical transducer which is set up to convert the sound signal from the environment into a corresponding electrical or electromagnetic signal, for example a microphone.
  • An output transducer generally includes an electro-acoustic transducer which is set up to generate an output sound signal from an electrical and / or electro-magnetic signal, for example a loudspeaker or a sound generator for bone sound conduction.
  • Signal processing is to be understood here in particular as a processing of the input signal or a signal derived from the input signal on the basis of user-specific requirements, i.e. in particular a frequency band-dependent amplification and / or noise suppression, the respective amplification factors in the individual frequency bands for correcting a possible hearing loss for the user Audiogram are designed.
  • Generation of the first intermediate signal as a function of the input signal is to be understood here in particular to mean that the signal processing directly receives the input signal as an input variable and generates the first intermediate signal therefrom, or that the signal processing receives a signal that is directly dependent on the input signal and generates the first intermediate signal therefrom, Thus, for example, the input signal, which was corrected by a compensation signal to compensate for acoustic feedback.
  • a classification of a hearing situation is to be understood in particular as meaning that groups of respectively similar acoustic environments, in which the user can be expected to find themselves, can be typed on the basis of measurable acoustic parameters, and that in particular depending on this typing, settings on the hearing aid and / or signal processing can be.
  • Listening situations include, for example, a conversation without background noise, a conversation with background noise, listening to music, driving in the car, several conversations overlaid with significant background noise at the same time (so-called "cocktail party” listening situation), etc.
  • a classification based on the input signal is to be understood in particular as a classification which directly uses the input signal itself as a relevant quantity, or a signal which is directly dependent on the input signal and which reproduces signal changes in the input signal in a comparable manner, for example the input signal corrected by a compensation signal, as a relevant quantity ,
  • a frequency shift is particularly considered as frequency distortion, which shifts the first intermediate signal by a predetermined amount in a predetermined frequency range.
  • the frequency range in which the shift is to be used and the amount of the shift are to be specified as parameters of the frequency distortion.
  • the frequency distortion can also be given by a frequency transposition with a more complex dependency between input frequency and output frequency.
  • the tonality or the volume of the sound signal of the environment can be determined in particular on the basis of the definitions customary for these parameters in psychoacoustics, the stationarity for example on the basis of the autocorrelation function of the input signal or its level variance, in each case over a suitably selected time window.
  • the method proposes three different dependencies, at least specifying a parameter of the frequency distortion: If only the hearing situation is classified on the basis of the input signal, the at least one parameter of the frequency distortion is also only specified as a function of the classification of the hearing situation. If at least one of the four parameters tonality, volume, stationarity and reverberation time is determined for the sound signal of the environment, then the at least one parameter of the frequency distortion becomes dependent only on at least one of these parameters specified.
  • the at least one parameter of the frequency distortion can be specified on the basis of this complete information, or for example only on the basis of the parameters mentioned for the sound signal the environment if the hearing situation is classified only for the setting of the signal processing.
  • the invention takes advantage of the fact that the first intermediate signal derived from the input signal, which usually also has a high degree of correlation with the input signal even after signal processing, is decorrelated by the frequency distortion from the input signal in the corresponding distorted frequency ranges. and such a decorrelation now leads to a considerable suppression of comb filter effects due to the resulting loss of coherence with the surrounding sound signal.
  • Comb filter effects occur precisely through an acoustic superimposition of the sound signal in the environment with an output sound signal generated by the output transducer if there is a fixed phase relationship between the superimposed signals. However, this fixed phase relationship is now broken up by the frequency distortion.
  • the method also takes into account that comb filter effects are not perceived by the user to be equally uncomfortable for any sound signals in the environment. Rather, for example, a comb filter effect and the resulting constructive and destructive interference at certain frequencies artificially generate a kind of overtone spectrum in a broadband, atonal sound signal, which leads to a quasi-tonal perception of the actually broadband sound signal, which can be perceived as unpleasant.
  • frequency distortion for example in the form of a frequency shift, in the case of very tonal sound signals, in particular in music, can lead to beats between the output sound signal of the hearing aid with frequency-shifted signal components and the direct sound signal from the surroundings, which is also perceived as very unpleasant can be, whereas comb filter effects with particularly tonal signals usually have no major effects on hearing.
  • the invention now opens up the possibility of making a decision in a simple manner simply on the basis of the hearing situation and / or of parameters of the sound signal that are easy to determine whether and to what extent it is at all probable that comb filter effects are formed in the present case, and how this threatens to impair the user's hearing, that is whether and how the frequency distortion should be adjusted to suppress the comb filter effects.
  • tuning the frequency distortion to at least one of the four parameters tonality, volume, stationarity and reverberation time of the sound signal of the environment allows a particularly detailed adaptation of the frequency distortion to the sound signal of the environment with regard to the expected perception of the output sound signal by the user.
  • the at least one parameter of the frequency distortion is preferably additionally specified as a function of a total gain and / or a frequency band-dependent gain factor of the signal processing.
  • the inclusion of the signal processing in the tuning of the frequency distortion by means of the at least one parameter offers the advantage, in particular frequency bands to be able to determine in which the formation of comb filter effects as a result of the respective increase or decrease are particularly likely or unlikely.
  • a comb filter parameter is determined on the basis of the classification of the hearing situation or on the basis of the at least one of the four parameters tonality, volume, stationarity and reverberation time, which parameter indicates a probability value for an occurrence and / or an intensity of a comb filter effect, the at least one parameter of the frequency distortion is additionally specified as a function of the comb filter parameter. If only one hearing situation is classified or only at least one of the parameters mentioned for the sound signal of the environment is determined, the comb filter parameter is determined in accordance with the information available in each case. If both the hearing situation is classified and the least one of the four parameters mentioned for the sound signal in the environment is determined, the comb filter parameter is preferably determined as a function of the complete information available.
  • the comb filter parameter can in particular be determined iteratively, a preliminary value being initially specified for the at least one parameter of the frequency distortion, and on the basis of this provisional value together with the further information available, the comb filter parameters when the frequency distortion is applied with the provisional one Value is determined.
  • the final value for the at least one parameter of the frequency distortion is then specified as a function of this comb filter parameter determined in this way.
  • the specification of the at least one frequency distortion parameter as a function of a comb filter parameter obtained in this way makes it possible, based on the available probability and potential intensity of a comb filter effect, preferably resolved via individual frequency bands, to carry out the specification in an optimization process which also depends on further parameters or parameters ,
  • An output signal is preferably generated by applying the predetermined frequency distortion to the first intermediate signal, the output signal being converted into an output sound signal by at least one output converter.
  • the output of the frequency-distorted first intermediate signal as an output signal, which is converted directly into the output sound signal, has the advantage that no further subsequent processes have to be considered for an optimal determination of the frequency distortion.
  • the at least one parameter of the frequency distortion is expediently additionally specified as a function of an expected acoustic superposition of frequency-distorted signal components of the output sound signal with the sound signal of the environment.
  • frequency-distorted signal components of the output sound signal can also impair the hearing perception of the user when acoustically overlaid with the sound signal of the environment , B. with a frequency shift as frequency distortion in the form of a beat between the only slightly frequency-shifted signal components of the output sound signal and the sound signal of the environment.
  • the at least one parameter of the frequency distortion is specified as a provisional value based on the classification of the listening situation and / or the least one of the four parameters tonality, volume, stationarity and reverberation time of the sound signal in the environment, and the information available shows that as a result If a high degree of tonality and / or stationarity of the sound signal is expected to produce a clearly perceivable beat, this can be taken into account when specifying the at least one parameter of the frequency distortion, and the frequency distortion occurs only for a few frequency ranges and / or with a lower intensity be completely turned off.
  • the at least one parameter of the frequency distortion is additionally dependent on an expected superimposition of frequency-distorted signal components of the first intermediate signal with non-frequency-distorted signal components of the first intermediate signal in the output signal.
  • the frequency distortion is given by a frequency shift, which is only to be applied to certain frequency bands, the finite edge steepness of the frequency band filter at a respective division frequency can result in a superposition of frequency-shifted signal components with signal components without frequency shift in the output signal. This can lead to unpleasant artifacts, especially in the case of tonal sound signals or in the event that there is considerable signal energy in the area of a division frequency.
  • the frequency response of the input signal can thus be taken into account accordingly when specifying the at least one parameter of the frequency distortion, and corresponding transitions between frequency ranges in which the frequency distortion is used and frequency ranges without frequency distortion can be specified such that a relatively low signal energy is present at the transitions to prevent artifacts from forming at the transitions.
  • the at least one parameter of the frequency distortion is predetermined by a function of a change in an output frequency as a function of an input frequency.
  • the frequency distortion can be characterized in a particularly comprehensive manner, and in particular can be tuned particularly precisely to the present acoustic situation.
  • a frequency shift is advantageously used as the frequency distortion, the at least one parameter being predetermined by the carrier of the function and / or the value of the frequency shift.
  • a frequency shift as frequency distortion can be implemented particularly easily in the form described, since the frequency range in which the frequency shift is to be used is to be implemented only by means of a filter, and the frequency shift takes place by a constant amount.
  • the frequency range in which the frequency shift is to be applied is the carrier of the function, coherent or semi-open, so that the said filter process can be implemented without any significant additional effort.
  • the frequency-distorted first intermediate signal is added to a feedback loop, a second intermediate signal being derived from the frequency-distorted first intermediate signal in the feedback loop, and the second intermediate signal being added to the input signal to suppress acoustic feedback.
  • the frequency-distorted first intermediate signal includes in particular the complete resulting signal after applying the frequency distortion to the first intermediate signal, that is to say including all frequency-distorted and non-frequency-distorted signal components.
  • the acoustic feedback takes place in particular by coupling an output sound signal of the hearing aid into the input transducer, so that signal components of the output sound signal are amplified again by the signal processing.
  • the signal resulting from the input signal and the second intermediate signal is preferably fed directly to the signal processing.
  • the occurrence of acoustic feedback is a general, frequently recurring problem for hearing aids, with the second sound signal preferably having to be decorrelated from the input signal in order to better suppress the acoustic feedback, especially in the case of particularly tonal sound signals, in order to prevent the formation of artifacts.
  • Such decorrelation can be achieved in particular through the present frequency distortion.
  • the at least one parameter of the frequency distortion is preferably additionally specified as a function of the acoustic feedback to be suppressed. If, for example, there is a particularly tonal sound signal, it may be advantageous for the user's hearing sensation to choose the area of application and / or the intensity of the frequency distortion rather low, and the comb filter effects often not perceived as critical in the case of tonal signals to take. However, if acoustic feedback now occurs, it can still go against the actual specifications, which were made according to the classification of the hearing situation and / or for the sound signal of the environment frequency distortion may be advantageous in order to be able to suppress acoustic feedback particularly effectively, since whistling tones which would otherwise occur would be even more disadvantageous for the hearing sensation of the user.
  • a signal signal corresponding to an audio signal is received by a signal receiver of the hearing device, the at least one parameter of the frequency distortion additionally being specified as a function of the audio signal.
  • a signal receiver includes in particular an antenna device which is set up to receive an electromagnetic transmission signal and a so-called “telecoil” which is set up to receive an inductive transmission signal.
  • the data signal corresponding to an audio signal in particular an electromagnetic or inductive signal is included, in which the audio signal is encoded in accordance with a corresponding protocol, so that after the data signal has been received by the signal receiver and after a subsequent decoding of the data signal, the acoustic information of the audio signal in the hearing device is available. This can be done in particular by a streaming signal from a consumer electronics device, e.g. via Bluetooth or the like, the case.
  • the at least one parameter of frequency distortion as an additional function of the audio signal, in particular at least one of the four parameters tonality, volume, stationarity and reverberation time and / or a time difference between the audio signal and the input signal can be determined for the audio signal. If, for example, when using television using a streaming signal from the television for the hearing aid while simultaneously using the loudspeakers of the television, it is found that the audio signal coded in the streaming signal has only minor tonal components at a time, the parameters of the frequency distortion are set to between the audio signal and the input signal containing the loudspeaker signal of the television, the signal delay occurring is coordinated, for example via an amount and a range of application of a frequency shift as frequency distortion.
  • the parameters of the frequency distortion can be changed accordingly, taking into account the fact that the input signal encodes many other sound signals in addition to that in the streaming signal or generally in a data signal Audio signal can include, for example Background noise, which can lead to comb filter effects in the hearing aid in the manner described above, regardless of the audio signal.
  • the invention further specifies a hearing aid with at least one input converter, at least one output converter and a control unit which is set up to carry out the prescribed method.
  • the advantages specified for the method and its further developments can be analogously transferred to the hearing aid.
  • FIG. 1 the frequency response for a direct sound signal 2 (dashed line), for an amplified output sound signal 4 of a hearing aid (dotted line) and a superimposed sound signal 6 (solid line) is shown schematically in a diagram, in which the sound level P is plotted against a frequency f .
  • the direct sound signal 2 is here by an in FIG. 1 Hearing aid, not shown, is amplified in a user-specific manner and output by the output transducer of the hearing aid as output sound signal 4.
  • the direct sound signal 2 and the output sound signal 4 are superimposed with a time delay.
  • the time-delayed superimposition leads to constructive interference 8, which overall leads to an increased sound level in the superimposed sound signal 6.
  • the time-delayed superimposition leads to destructive interference 10, which sometimes even results in an almost complete cancellation in the superimposed sound signal 6.
  • the maxima for the constructive interferences 8 are found in each case at integer multiples of that frequency which corresponds to the reciprocal time delay in the hearing device, the minima of the destructive interferences 10 in each case at half-number multiples of this frequency.
  • the user-specific amplification for generating the output sound signal 4 and the time delay that occurs the comb filter effects that occur can be perceived by the user of the hearing aid as very unpleasant.
  • FIG. 2 a method 20 is shown schematically in a block diagram, by means of which a negative hearing sensation caused by comb filter effects is to be prevented as far as possible for the user during operation of a hearing device 22.
  • parameters 30 are determined on the basis of the input signal 26, by means of which statements about the tonality, volume, stationarity and reverberation time are possible for the sound signal 2.
  • the input signal 26 is fed to a signal processing unit 32, in which the user-specific signal processing 34 customary for the hearing aid 22 takes place on the basis of the audiological requirements of the user.
  • the signal processing 34 includes in particular a breakdown of the input signal 26 into different frequency bands, an amplification of the input signal 26 with frequency band-dependent gain factors, and frequency band-dependent processes for noise suppression, which can also depend on the classification 28 of the listening situation.
  • the signal processing unit 32 now outputs a first intermediate signal 36, to which frequency distortion 38 is applied, as will be described below.
  • an output signal 40 is generated which is converted into an output sound signal 4 by an output converter 42 of the hearing device 22.
  • the output converter 42 is given by a loudspeaker.
  • the hearing aid 22 has a signal receiver 43 for receiving a data signal 44, in which an audio signal 45 is encoded.
  • the signal receiver 43 can be given, for example, by an antenna device, the data signal 44, for example, by a Bluetooth signal.
  • the audio signal 45 can also be decoded from the data signal 44 by a processor of the signal receiver 43 that is set up for this purpose.
  • the audio signal 45 can also first be extracted from the data signal 44 in the signal processing unit 32 be decoded.
  • the audio signal 45 if present, is processed by the signal processing unit 32 and is included in the first intermediate signal 36.
  • the output signal 40 is also branched off into a feedback loop 48.
  • a second intermediate signal 52 is derived from the output signal 40 by an adaptive filter 50, which is fed to the input signal 26 to compensate for the acoustic feedback 46.
  • the input signal 26 compensated for by the second intermediate signal 52 is fed as an error signal 54 to the adaptive filter 50 as a further input variable.
  • the frequency distortion 38 is given by a frequency shift which constantly shifts the first intermediate signal 36 by a fixed amount ⁇ above a division frequency ft.
  • a comb filter parameter 56 is first determined on the basis of the classification 28 of the hearing situation and the parameters 30 via the tonality, volume, stationarity and reverberation time of the sound signal 2 of the environment, which parameter is present in the present Listening situation and the available parameters 30 indicates the probability of the occurrence of a comb filter effect and its possible intensity.
  • the frequency shift decorrelates the output signal 4 from the sound signal 2 of the environment, which in principle suppresses the formation of comb filter effects.
  • This decorrelating effect can also have effects on the suppression of the acoustic feedback 46, which is why the division frequency ft and the amount ⁇ of the displacement can also be predetermined as a function of the acoustic feedback 46 to be suppressed, for example by corresponding correlation measurements in the adaptive filter 50 for determining the division frequency ft and the amount ⁇ of the shift of the frequency shift the parameters 30, which characterize the sound signal 2 of the environment, are also included in such a way that possible beats which can occur between the sound signal 2 and the output sound signal 4 are also taken into account.
  • signal-internal superimpositions of signal components, to which the frequency shift has been applied, with such signal components, which consist of the unchanged first intermediate signal 36, can also be taken into account in the output signal 40.
  • the division frequency ft should preferably be set such that such superimposition of the signal components due to the finite steepness of the filters used has the least possible effects in the output signal 40. This can be achieved, for example, by placing the division frequency ft in a frequency band with a particularly low signal energy.
  • the final determination of the division frequency ft and the amount ⁇ of the shift can then be carried out in an optimization process of several variables, which are based on the present listening situation, the sound properties of the sound signal 2 determined by the parameters 30, a possible acoustic feedback 46 and possible superimposition of the individual signal components is to be carried out with corresponding priority.
  • efficient suppression of acoustic feedback 46 can first be given the highest priority, and then, depending primarily on the hearing situation and the tonality, which was determined for sound signal 2, the frequency shift can be set such that beatings are possible for particularly tonal sound signals 2 are to be avoided and the frequency shift is correspondingly smaller, while comb filter effects are to be avoided for particularly broadband, atonal signals, and accordingly the division frequency ft should already be selected in a low frequency range.
  • the final determination of the division frequency ft can then be determined as a function of the signal energies of individual frequency bands of the already predetermined frequency range, in order to minimize the effects of overlapping frequency-shifted signal components with non-frequency-shifted signal components in the output signal.
  • FIG. 3 is comparable to that FIG. 1 , the frequency response for the direct sound signal 2 (dashed line), for the output sound signal 4 (dotted line) and for the superimposed sound signal 6 (solid line).
  • the method 20 was followed when the output sound signal 4 was formed FIG. 2 applied. It can now be seen that the relatively broadband atonal sound signal 2 no longer leads to the occurrence of comb filter effects when superimposed on the output sound signal 4.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)

Claims (11)

  1. Procédé (20) de fonctionnement d'un appareil auditif (22) qui comprend au moins un transducteur d'entrée (24) et au moins un transducteur de sortie (42), un signal d'entrée (26) étant généré par l'au moins un transducteur d'entrée (24) à partir d'un signal sonore (2) de l'environnement,
    une classification (28) d'une situation auditive de l'environnement étant effectuée sur la base du signal d'entrée (26) et/ou l'un au moins des quatre paramètres (30) tonalité, volume, stationnarité et temps de réverbération étant déterminé pour le signal sonore (2) de l'environnement,
    un premier signal intermédiaire (36) étant généré en fonction du signal d'entrée (26) à l'aide d'un processeur de signal (32),
    au moins un paramètre (ft, Δ) d'une distorsion de fréquence (38) étant spécifié sur la base de la classification (28) de la situation auditive ou sur la base de l'un au moins des quatre paramètres (30) tonalité, volume, stationnarité et temps de réverbération, et
    la distorsion de fréquence (38) spécifiée étant appliquée au premier signal intermédiaire (36),
    caractérisé en ce que
    - un paramètre de filtre en peigne (56), qui indique une valeur de probabilité d'occurrence et/ou d'intensité d'un effet de filtre en peigne, est déterminé sur la base de la classification (28) de la situation auditive ou sur la base de l'un au moins des quatre paramètres (30) tonalité, volume, stationnarité et temps de réverbération, et
    - l'au moins paramètre (ft, Δ) de la distorsion de fréquence (38) est également spécifié en fonction du paramètre de filtre en peigne (56).
  2. Procédé (20) selon la revendication 1,
    l'au moins un paramètre (ft, Δ) de la distorsion de fréquence (38) étant en outre spécifié en fonction d'un gain global et/ou d'un facteur d'amplification du traitement de signal (32) qui dépend de la bande de fréquence.
  3. Procédé (20) selon l'une des revendications précédentes,
    un signal de sortie (40) étant généré par l'application de la distorsion de fréquence (38) spécifiée au premier signal intermédiaire (36) et
    le signal de sortie (40) étant converti en un signal sonore de sortie (4) par l'au moins un transducteur de sortie (42).
  4. Procédé (20) selon la revendication 3,
    l'au moins un paramètre (ft, Δ) de la distorsion de fréquence (38) étant en outre spécifié en fonction de la superposition acoustique attendue de composantes de signal distordues en fréquence du signal sonore de sortie (4) avec le signal sonore (2) de l'environnement.
  5. Procédé (20) selon la revendication 3 ou 4,
    l'au moins un paramètre (ft, Δ) de la distorsion de fréquence (38) est en outre spécifié en fonction de la superposition attendue de composantes de signal distordues en fréquence du premier signal intermédiaire (36) avec des composantes de signal non distordues en fréquence du premier signal intermédiaire (36) dans le signal de sortie (40).
  6. Procédé (20) selon l'une des revendications précédentes,
    l'au moins un paramètre (ft, Δ) de la distorsion de fréquence (38) étant donné par une fonction de la variation de la fréquence de sortie en fonction d'une fréquence d'entrée.
  7. Procédé (20) selon la revendication 6,
    un décalage de fréquence étant appliqué en tant que distorsion de fréquence (38), l'au moins un paramètre (ft, Δ) étant donné par la porteuse de la fonction et/ou la valeur (Δ) du décalage de fréquence.
  8. Procédé (20) selon l'une des revendications précédentes,
    le premier signal intermédiaire (40) distordu en fréquence étant amené à une boucle de réaction (48),
    un deuxième signal intermédiaire étant dérivé du premier signal intermédiaire (40) distordu en fréquence dans la boucle de réaction (48) et
    le deuxième signal intermédiaire (52) étant ajouté au signal d'entrée (26) pour supprimer la rétroaction acoustique (46).
  9. Procédé (20) selon la revendication 8,
    l'au moins un paramètre (ft, Δ) de la distorsion de fréquence (38) étant en outre spécifié en fonction de la rétroaction acoustique (46) à supprimer.
  10. Procédé (20) selon l'une des revendications précédentes,
    un signal de données (44) qui correspond à un signal audio (45) étant reçu par un récepteur de signal (43) de l'appareil auditif (22), et
    l'au moins un paramètre (ft, Δ) de la distorsion de fréquence (38) étant en outre spécifié en fonction du signal audio (45).
  11. Appareil auditif (22) comprenant au moins un transducteur d'entrée (24), au moins un transducteur de sortie (42) et une unité de commande qui est conçue pour mettre en œuvre le procédé (20) selon l'une des revendications précédentes.
EP17207540.0A 2016-12-22 2017-12-15 Procédé de fonctionnement d'un dispositif de correction auditive Active EP3340656B1 (fr)

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DE102018207780B3 (de) 2018-05-17 2019-08-22 Sivantos Pte. Ltd. Verfahren zum Betrieb eines Hörgerätes
DE102022204349A1 (de) 2022-05-03 2023-11-09 Sivantos Pte. Ltd. Verfahren zum Betrieb eines Hörgerätes und Hörgerät
DE102024206995A1 (de) * 2024-07-25 2026-01-29 Sivantos Pte. Ltd. Verfahren zum Betrieb eines Hörgeräts

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JP4939935B2 (ja) * 2003-06-24 2012-05-30 ジーエヌ リザウンド エー／エス 整合された音響処理を備える両耳用補聴器システム
AU2005232314B2 (en) 2005-11-11 2010-08-19 Phonak Ag Feedback compensation in a sound processing device
DK2064918T3 (en) * 2006-09-05 2015-01-26 Gn Resound As A hearing-aid with histogram based lydmiljøklassifikation
DK2369859T3 (en) * 2008-05-30 2017-03-13 Sonova Ag Method of adapting sound in a hearing aid by frequency change and such a device / Method of adapting sound in a hearing aid device by frequency modification and such a device
WO2010088960A1 (fr) 2009-02-06 2010-08-12 Oticon A/S Substitution de bande spectrale pour éviter des sifflements d'accrochage et une sous-oscillation
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DE102011075006B3 (de) * 2011-04-29 2012-10-31 Siemens Medical Instruments Pte. Ltd. Verfahren zum Betrieb eines Hörgerätes mit verringerter Kammfilterwahrnehmung und Hörgerät mit verringerter Kammfilterwahrnehmung
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EP2670168A1 (fr) * 2012-06-01 2013-12-04 Starkey Laboratories, Inc. Dispositif d'assistance auditive adaptatif utilisant la détection et la classification d'environnement multiple
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CN108235210B (zh) 2020-11-17
DE102016226112A1 (de) 2018-06-28
US10652670B2 (en) 2020-05-12
CN108235210A (zh) 2018-06-29
US20180184217A1 (en) 2018-06-28
DK3340656T3 (en) 2020-04-27
EP3340656A1 (fr) 2018-06-27

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