EP2802158B1 - Procédé d'adaptation du signal utile dans des systèmes d'aide à l'audition binauraux - Google Patents

Procédé d'adaptation du signal utile dans des systèmes d'aide à l'audition binauraux Download PDF

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Publication number
EP2802158B1
EP2802158B1 EP14155511.0A EP14155511A EP2802158B1 EP 2802158 B1 EP2802158 B1 EP 2802158B1 EP 14155511 A EP14155511 A EP 14155511A EP 2802158 B1 EP2802158 B1 EP 2802158B1
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Prior art keywords
signal
audio signals
predetermined
hearing aid
filter
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German (de)
English (en)
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EP2802158A3 (fr
EP2802158A2 (fr
Inventor
Walter Kellermann
Stefan Meier
Klaus Reindl
Marc Aubreville
Eghart Fischer
Homayoun Kamkar Parsi
Henning Puder
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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; 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
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • 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
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/407—Circuits for combining signals of a plurality of transducers
    • 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
    • H04R25/55—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception using an external connection, either wireless or wired
    • H04R25/552—Binaural
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility

Definitions

  • the invention relates to a method for operating a hearing aid system and a hearing aid system with at least two hearing aid devices, between which a signal path is provided, and with at least one signal processing unit which is provided for processing audio signals.
  • hearing impairment affects both ears
  • the hearing impaired should be provided binaurally with hearing aids.
  • Modern hearing aids have signal processing algorithms which automatically vary the parameters of the hearing aids as a function of the hearing situation. In the case of binaural care, the hearing situation is assessed on both ears.
  • Noise and noise are omnipresent in everyday life and make speech communication more difficult, especially if there is an impairment of natural hearing. Therefore, techniques are desirable that suppress noise and noise, but change the desired sounds and tones, hereinafter also referred to as useful signals, as little as possible.
  • the known embodiments of techniques for suppressing noise and noise require for effective suppression of noise noises with identical as possible Nutzsignalan tone.
  • the various signals are detected by a plurality of different microphones, which are arranged on the two hearing aids.
  • the useful signals detected by these microphones differ substantially due to ambient reverberation and the acoustic influence of the head of the wearer.
  • downstream noise suppression techniques do not work effectively and especially when it comes to adaptive methods, they converge only very slowly or not at all. This leads to a poor suppression of the noise and additional acoustic artifacts by the noise reduction.
  • a method for blind source separation by means of Wiener filter is from the patent application EP 2211563 A1 known.
  • a hearing aid system with two hearing aids in which input signals by means of a first signal channel with a first filter, and with a second signal channel with a second filter are converted into output signals.
  • the coefficients of at least one of the filters are adjusted so that the difference between the output signals of the first and second signal channels is minimized.
  • the WO 2009/081193 A1 relates to a system for noise suppression, comprising a digital filter with a predetermined filter and with an adaptive filter, and with a signal processing device for generating a control signal for the adaptive filter.
  • the object of the present invention is to provide a method for operating a hearing aid system and a hearing aid system, by which the noise suppression in binaural hearing aid systems is improved.
  • This object is achieved by a method having the features of claim 1.
  • the part of the object relating to the hearing aid system is achieved by the features of claim 12.
  • the method according to the invention relates to a method for operating a hearing aid system with at least two hearing aid devices.
  • the hearing aids have a transducer for receiving an acoustic signal and conversion into a respective first audio signal.
  • the hearing aid system further comprises a signal processing device for processing audio signals, as well as a signal connection for transmitting a first audio signal from each hearing aid device to the signal processing device.
  • the acoustic signal has a useful signal, which arrives from a given spatial direction with respect to the hearing aid system.
  • the signal processing device filters the first audio signals with a filter predetermined for the given spatial direction, so that second audio signals are generated, wherein the predetermined filter is designed such that the components of the useful signal in the second audio signals are matched to one another to a greater degree than in the first audio signals; and Filtering the second audio signals with an adaptive filter, so that third audio signals are generated, wherein the adaptive filter is designed such that the portions of the useful signal in the third audio signals are aligned to a higher degree than in the second audio signals.
  • the method according to the invention provides second audio signals in which the components of the useful signal are matched to one another to a greater degree than in the first audio signals.
  • the adaptive filter of the second step converges much faster and therefore provides third audio signals in which the alignment of the useful signals is further improved. This allows subsequent noise suppression methods from these third audio signals to form better reference signals for the useful and interfering signals and noise almost completely to suppress.
  • the method of the invention allows shorter adaptive filters with less computational power requirement and faster convergence of the adaptive filter, which advantageously allows adaptation to rapidly changing environmental conditions.
  • the hearing aid system according to the invention for carrying out the method according to the invention shares its advantages.
  • the signal processing device has means for dividing the first audio signals into first audio signals in predetermined frequency ranges, wherein the steps of the method according to the invention are performed separately for each audio signal in a frequency range.
  • the predetermined filter is defined by a predetermined number of predetermined coefficients.
  • a filter with a predetermined coefficient number is easy to implement and advantageously requires only a small memory space.
  • the predetermined number of the predetermined coefficients is different in different frequency ranges.
  • the need for computing power for the predetermined filter can be optimized in an advantageous manner by more accurate filtering takes place only in frequency ranges with corresponding noise.
  • the memory requirement for the filter is further reduced.
  • predetermined coefficients for the predetermined filter are predetermined for at least two predetermined spatial directions.
  • this further comprises a step of estimating a spatial direction of the useful signal, wherein in the step of filtering the first audio signals, the predetermined filter is predetermined by the coefficients whose assigned predetermined spatial direction is closest to the estimated spatial direction.
  • the predetermined filter can be set to different predetermined spatial directions, wherein in each case the selected spatial direction of the spatial direction of the source of the useful signal is closest, so that the predetermined filter achieves an optimal filtering effect.
  • a reference signal for a noise is derived from the second and / or third audio signals by subtraction.
  • a reference signal for a useful signal is derived from the second and / or third audio signals by summation.
  • the reference signal is used for a useful signal for detecting a voice activity.
  • the reference signal for the useful signal for example, when used to detect a voice activity (VAD, voice activity detection) allows a safer detection.
  • the predetermined filter is a finite impulse response (FIR) filter.
  • FIR finite impulse response
  • An FIR filter advantageously has no tendency to oscillate and thus ensures a convergence of a provided signal under all circumstances.
  • Fig. 1 shows the basic structure of a hearing aid system 100 according to the invention.
  • the hearing aid system 100 has two hearing aid devices 110, 110 '.
  • a hearing aid housing 1, 1 ' for carrying behind the ear, one or more microphones 2, 2' for receiving the sound or acoustic signals from the environment are installed.
  • a signal processing unit 3, 3 ' which is also integrated in the hearing aid housing 1, 1', processes the first audio signals.
  • the output signal of the signal processing unit 3, 3 ' is transmitted to a loudspeaker or receiver 4, 4', which outputs an acoustic signal.
  • the sound is optionally transmitted via a sound tube, which is fixed with an earmold in the ear canal, to the eardrum of the device carrier.
  • the power supply of the hearing device and in particular of the signal processing unit 3, 3 ' takes place by means of a likewise integrated into the hearing aid housing 1, 1' battery 5, 5 '.
  • the hearing aid system 100 has a signal connection 6, which is designed to transmit a first acoustic signal from the signal processing device 3 to the signal processing device 3 '. It is provided in the preferred embodiment that also signal processing device 3 'transmits a first acoustic signal to the signal processing device 3 in the opposite direction. Furthermore, it is conceivable to transmit the signals of several or all of the microphones 2, 2 'in each case to the other hearing aid device 110, 110'.
  • a signal connection 6 are wired, optical or wireless connections such. Bluetooth conceivable.
  • Fig. 2 shows a schematic flow diagram of a method according to the invention in the signal processing device 3, 3 '.
  • a spatial direction of a desired signal is estimated by the signal processing device 3, 3 '. This can be done for example by time or phase differences for a prominent signal in the first audio signals. An analysis of the amplitude of this distinctive signal in the first audio signals is also conceivable. From the estimated spatial direction of the useful signal, the signal device selects a spatial direction from a set of predetermined spatial directions which is closest to the estimated spatial direction of the useful signal. If no distinctive signal is present, an estimate of the spatial direction is used at an earlier time or a predetermined direction, e.g. selected in a straight line in front of the face of the wearer of the hearing aid system. It is also conceivable to undertake a new estimation of the spatial direction only in the case of a multiple occurrence of a distinctive signal having the same characteristics.
  • step S20 the first audio signals are filtered by the predetermined filter 31.
  • the predetermined filter 31 uses predetermined filter coefficients for the given one Spatial direction of the wanted signal. For example, for each element of the set of given spatial directions, a set of predetermined filter coefficients for the predetermined filter 31 may be stored in a look-up table.
  • the predetermined filter 31 provides second audio signals.
  • the useful signal is already equalized as a signal component by the use of the predetermined filter 31, ie, the differences in amplitude and phase of the useful signal are smaller between the individual second audio signals than between the corresponding first ones Audio signals. Under real ambient conditions, a compensation in the range of 5 - 10 dB can be achieved.
  • the adaptive filter 32 filters the second audio signals.
  • an adaptive filter for example, the Least Mean Square (LMS) method described by Teutsch and Elko can be used to minimize the noise. Due to the pre-filtering by the predetermined filter 31, the filter effect by the adaptive filter 32 may be smaller. Therefore, for example, a filter with fewer coefficients can be used, and therefore an algorithm for adaptively adapting the filter coefficients to the second audio signals converges faster.
  • LMS Least Mean Square
  • the adaptive filter 32 provides third audio signals.
  • the useful signal is already adjusted by the application of the adaptive filter 32 as a signal component to an increased extent, that is, the differences in amplitude and phase of the useful signal between the individual third audio signals are even lower than between the corresponding second audio signals.
  • a reference signal for a noise can be formed in a conceivable embodiment by subtracting third audio signals, which is used in a step S50 by a noise suppression method is to suppress noise in the audio signals before they are output via the handset 4 as acoustic signals. Due to the filtering, the useful signal in the different parts of the third audio signal is almost identical, so that it almost completely extinguishes during difference formation. Therefore, artefacts which distort the wanted signal hardly occur in the subsequent noise suppression method.
  • a reference signal for the useful signal is generated. Since the components of the useful signal are matched by the filtering, but the disturbances are statistically distributed, addition of the useful signal to the interference is further emphasized. Therefore, the reference signal for the wanted signal obtained in step S60 has a better signal-to-noise ratio and can be used for voice actvity detection in a further step S70.
  • the steps S10 to S70 may be applied to audio signals having a single continuous frequency range.
  • the first audio signals are first divided into different frequency ranges before all or individual steps S10 to S70 are applied to the individual frequency ranges.
  • This has the advantage of using a different gain and other signal processing, such as a dynamic compression in a hearing aid to compensate for a hearing loss of the patient for individual frequency ranges. Means for dividing the audio signals into frequency ranges are therefore usually already present and the steps S10 to S70 can be easily incorporated into the existing processing process.
  • individual steps such as, for example, step S10 for estimating a given spatial direction, take place jointly for all frequencies.
  • Fig. 3 shows a corresponding schematic representation of a hearing aid system 100 according to the invention with a frequency distribution.
  • FIG. 3 On the left is in Fig. 3 the head 200 of a patient is shown with an array of two microphones 2, 2 'on each side of the head. The other functional blocks are then no longer specified in their relative position to the head.
  • the first audio signals of the microphones 2, 2 ' are supplied via a signal connection 6 to both hearing aid devices 110, 110' and the signal processing device 3, 3 '.
  • each of the four first audio signals in each hearing aid device 110, 110 ' is divided into a plurality of frequency bands.
  • the signal processing device 3, 3 ' carries out the step S20, which is represented by the function block of the predetermined filter 31, 31'.
  • the second audio signals are supplied to the adaptive filter 32, 32 ', which performs step S30 on the individual frequency ranges of the second audio signals and provides third audio signals.
  • synthesis of the individual frequency ranges of the third audio signals takes place in the synthesis filter bank 33, 33', so that four third audio signals each comprise a single, at least parts of the audible range Frequency range arise.
  • the analysis filter bank 10, 10 ' which divides the first audio signals into frequency ranges, can be realized for example by a short-time Fourier transformation.
  • the further processing of the second and third audio signals then takes place in the short-time Fourier range, before they are in the synthesis filter bank 33, 33 'in turn assembled by, for example, an inverse short-term Fourier transform to form an audio signal.
  • the signal processing unit 3 configured to carry out steps S10 to S70, while the other signal processing unit 3' is for providing the first audio signals and transmitting the audio signals via the signal connection 6 limited.
  • the energy consumption for a parallel processing of all signals in both hearing aids 110, 110 'could be avoided.
  • Other architectures, for example, with a central, remote signal processing unit 3 are conceivable.
  • Fig. 4 shows a schematic representation of a test setup for determining the direction-dependent predetermined filter coefficients.
  • the filters 31 are predetermined, for example by predetermined filter coefficients. These filter coefficients can either be theoretically determined by a model calculation, or determined by measurements and adaptive methods in the laboratory and stored in tables in the signal processing unit 3, 3 '.
  • an artificial head 200 eg KEMAR
  • a measuring device receives the audio signals of the microphones as a function of the spatial direction of a signal source 120 given by the angle ⁇ , which emits a useful signal as sound.
  • Modeling and optimization methods can then be used to determine a set of coefficients that are suitable for predetermined spatial directions under the idealized conditions the measurement equalizes the components of the useful signal in the second audio signals.
  • a spatial direction of a real useful signal source can later be estimated via the step S10, and in the predetermined filter 31 the corresponding coefficient set for a predetermined spatial direction closest to the estimated spatial direction can be selected and used for filtering. Since the real conditions deviate from the ideal conditions of the measurement, the matching of the components of the useful signal in the second audio signals is not completely achieved, which is why the method according to the invention provides for a subsequent adaptive filter 32, 32 '.

Claims (12)

  1. Procédé pour le fonctionnement d'un système d'aide auditive (100) comportant au moins deux appareils auditifs (110, 110'), dans lequel les appareils auditifs (110, 110') comportent un transducteur (2, 2') destiné à recevoir un signal acoustique et à le convertir en un premier signal audio respectif, dans lequel le système d'aide auditive (110, 110') comporte un dispositif de traitement de signaux (3, 3') destiné à traiter des signaux audio et le système d'aide auditive comporte une liaison de signal (6) destinée à transmettre le premier signal audio respectif de chaque appareil auditif (110, 110') au dispositif de traitement de signaux (3, 3'), et dans lequel l'appareil de traitement de signaux (3, 3') comporte un filtre prédéterminé (31, 31') ayant des coefficients de filtre prédéterminés pour une direction spatiale donnée,
    caractérisé en ce que le dispositif de traitement de signaux (3, 3') exécute les étapes suivantes pour ajuster les proportions d'un signal utile dans les signaux audio acquis du signal acoustique en des troisièmes signaux audio, le signal utile arrivant en provenance de ladite direction spatiale donnée par rapport au système d'aide auditive :
    filtrer (S20) les premiers signaux audio au moyen d'un filtre (31, 31') prédéterminé pour la direction spatiale donnée de manière à générer des deuxièmes signaux audio, le filtre prédéterminé (31, 31') étant conçu de manière à ce que les proportions du signal utile dans les deuxièmes signaux audio soient ajustées les unes par rapport aux autres dans une plus grande mesure que dans les premiers signaux audio ; et
    filtrer (S30) les deuxièmes signaux audio au moyen d'un filtre adaptatif (32, 32') de manière à générer des troisièmes signaux audio, le filtre adaptatif (32, 32') étant conçu de manière à ce que les proportions du signal utile dans les troisièmes signaux audio soient ajustées les unes par rapport aux autres dans une plus grande mesure que dans les deuxièmes signaux audio.
  2. Procédé selon la revendication 1, dans lequel le dispositif de traitement de signaux comporte des moyens (10, 10') destinés à répartir les premiers signaux audio en des premiers signaux audio dans des plages de fréquences prédéterminées, et en ce que les étapes de la revendication 1 sont réalisées séparément pour chaque signal audio dans une plage de fréquences.
  3. Procédé selon la revendication 1 ou 2, dans lequel le filtre prédéterminé (31, 31') est défini par un nombre prédéterminé de coefficients prédéterminés.
  4. Procédé selon les revendications 2 et 3, dans lequel le nombre prédéterminé de coefficients prédéterminés est différent dans des gammes de fréquences différentes.
  5. Procédé selon l'une des revendications précédentes, dans lequel des coefficients prédéterminés pour le filtre prédéterminé sont prédéterminés pour au moins deux directions spatiales prédéterminées.
  6. Procédé selon la revendication 5, dans lequel le procédé comporte en outre une étape consistant à estimer une direction spatiale du signal utile et, lors de l'étape de filtrage des premiers signaux audio, le filtre prédéterminé (31, 31') est prédéterminé par les coefficients dont la direction spatiale prédéterminée associée est la plus proche de la direction spatiale estimée.
  7. Procédé selon l'une des revendications précédentes, dans lequel un signal de référence pour un bruit parasite est dérivé des deuxièmes et/ou troisièmes signaux audio par calcul de différence.
  8. Procédé selon l'une des revendications précédentes, dans lequel un signal de référence pour un signal utile est dérivé des deuxièmes et/ou troisièmes signaux audio par sommation.
  9. Procédé selon la revendication 8, dans lequel le signal de référence pour un signal utile est utilisé pour reconnaître une activité vocale.
  10. Procédé selon l'une des revendications précédentes, dans lequel le filtre prédéterminé (31, 31') est un filtre à réponse impulsionnelle finie.
  11. Procédé selon l'une des revendications précédentes, dans lequel le filtre adaptatif (32, 32') utilise un procédé LMS (Least Mean Square) pour minimiser le bruit parasite.
  12. Système d'aide auditive (100) comportant au moins deux appareils auditifs (110, 110'), dans lequel les appareils auditifs (110, 110') comportent un transducteur (2, 2') destiné à acquérir un signal acoustique et à le convertir en un premier signal audio respectif, dans lequel le système d'aide auditive comprend un dispositif de traitement de signaux (3, 3') destiné à traiter des signaux audio et le système d'aide auditive comprend une liaison de signal (6) destinée à transmettre des premiers signaux audio respectifs de chaque appareil auditif (110, 110') au dispositif de traitement de signaux (3, 3'), dans lequel le dispositif de traitement de signaux (3, 3') est configuré pour mettre en oeuvre le procédé selon l'une des revendications 1 à 11.
EP14155511.0A 2013-04-19 2014-02-18 Procédé d'adaptation du signal utile dans des systèmes d'aide à l'audition binauraux Active EP2802158B1 (fr)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
DE102013207161.2A DE102013207161B4 (de) 2013-04-19 2013-04-19 Verfahren zur Nutzsignalanpassung in binauralen Hörhilfesystemen

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EP2802158A2 EP2802158A2 (fr) 2014-11-12
EP2802158A3 EP2802158A3 (fr) 2017-07-26
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US9277333B2 (en) 2016-03-01
DK2802158T3 (da) 2019-11-18
EP2802158A3 (fr) 2017-07-26
DE102013207161B4 (de) 2019-03-21
EP2802158A2 (fr) 2014-11-12
DE102013207161A1 (de) 2014-11-06
US20140314259A1 (en) 2014-10-23

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