EP3059979B1 - Prothèse auditive avec amélioration de signal - Google Patents

Prothèse auditive avec amélioration de signal Download PDF

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Publication number
EP3059979B1
EP3059979B1 EP16164585.8A EP16164585A EP3059979B1 EP 3059979 B1 EP3059979 B1 EP 3059979B1 EP 16164585 A EP16164585 A EP 16164585A EP 3059979 B1 EP3059979 B1 EP 3059979B1
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Prior art keywords
signal
estimate
hearing aid
noise
microphone
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EP16164585.8A
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German (de)
English (en)
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EP3059979A1 (fr
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Karl-Fredrik Johan GRAN
Andrew Burke Dittberner
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GN Hearing AS
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GN Hearing AS
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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
    • H04R25/55—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
    • 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

Definitions

  • a new binaural hearing aid system that compensates for a hearing impaired user's loss of ability to understand speech in noise.
  • Hearing impaired individuals often experience at least two distinct problems: a hearing loss, which is an increase in hearing threshold level, and a loss of ability to understand high level speech in noise in comparison with normal hearing individuals. For most hearing impaired patients, the performance in speech-in-noise intelligibility tests is worse than for normal hearing people, even if the audibility of the incoming sounds is restored by amplification.
  • An individual's speech reception threshold (SRT) is the signal-to-noise ratio required in a presented signal to achieve 50 percent correct word recognition in a hearing in noise test.
  • the patient's hearing ability can thus be improved by making previously inaudible speech cues audible.
  • At least one microphone in combination with a number of filters, fixed or adaptive, is used to enhance a signal from the presumed target direction and at the same time suppress all other signals.
  • the goal is to create an estimate of the long term noise spectrum and turn down gain in frequency bands where the instantaneous target signal power is lower than the long term noise power. Even though the methods are very different from a technological standpoint, they still have the common goal; enhance the target signal and remove the noise disturbance.
  • the methods cannot take listener intent into account and they may remove parts of the audio signal which the listener is trying to focus on.
  • WO 2011/006496 A1 discloses a hearing aid system with a processing unit for adaptive wind noise suppression.
  • the processing unit comprises a first microphone and a second microphone.
  • the analogue signal from the first microphone is converted to a first digital signal in a first A/D converter and the analogue signal from the second microphone is converted to a second digital signal in a second A/D converter.
  • the output of the first A/D converter is operationally connected to a first input of a subtraction node.
  • the output of the second A/D converter is operationally connected to the input of an adaptive filter.
  • the output of the adaptive filter is branched and in a first branch operationally connected to the second input of the subtraction node and in a second branch operationally connected to the input of the remaining signal processing in the hearing aid.
  • the output from the subtraction node is operationally connected to a control input of the adaptive filter.
  • a new method of enhancement of a desired signal is disclosed.
  • the new method makes use of the human auditory system's capability of concentrating on a desired signal.
  • a new binaural hearing aid system using the new method is also disclosed.
  • ITD interaural time differences
  • ILD interaural level differences
  • the level difference is a result of diffraction and is determined by the relative position of the ears compared to the source. This cue is dominant above 2 kHz but the auditory system is equally sensitive to changes in ILD over the entire spectrum.
  • At least one of the target signal and the noise signal is estimated, and the at least one estimate is presented to the user of the binaural hearing aid system in such a way that a user's capability of understanding speech in noise is improved.
  • a listener may listen to sound with a signal S that the listener desires to listen to and noise N that the listener finds disturbing, i.e. the sound signal is S+N.
  • the desired signal S may be estimated.
  • the estimate is denoted ES.
  • S+N-ES-ES Subtracting two times the estimate ES from the sound signal S+N results in a modified signal: S+N-ES-ES, and since ES is approximately equal to S, modified signal is: N-ES which is approximately equal to -S +N, i.e. the original sound signal wherein the desired signal S has been substantially substituted with signal S phase shifted by 180°.
  • the original signal S+N may be presented to one ear of a user, and the phase shifted signal N-ES, or more accurately S+N-2ES, may be presented to the other ear for improved BMLD and SRT.
  • both the desired signal S and the noise N may be estimated and the sum of the estimates ES+EN may be presented to one ear of the user, and the phase shifted sum -ES+EN may be presented to the other ear for improved BMLD and SRT.
  • the desired signal S and the noise may be swapped so that the noise estimated is phase shifted instead of the desired signal for improved BMLD and SRT; however with decreased performance compared to phase shifting the desired signal S.
  • Noise can be background speech, restaurant clatter, music (when speech is the desired signal), traffic noise, etc.
  • the purpose for the method is not to remove any part of the signal but instead present the signals so that the auditory system can perform natural noise reduction and separate the target signal from the noise signal.
  • the target signal and the noise signal are swapped; enhancement of the target signal is still obtained, although with slightly decreased performance.
  • a new binaural hearing aid system comprising at least one microphone for provision of respective at least one microphone audio signal in response to sound received at the at least one microphone, a signal separation unit configured to provide an estimate of one of a target signal and a noise signal based on the at least one microphone audio signal, a phase shift circuit configured to phase shift the estimate of one of the target signal and the noise signal, and a phase shift adder connected to provide a phase shifted signal representing sound received at the at least one microphone in which the estimate of one of the target signal and the noise signal has substantially substituted the respective original one of the target signal and the noise signal, and a first receiver for conversion of a receiver input signal into an acoustic signal for transmission towards one of the eardrums of a user of the binaural hearing aid system, and a second receiver for conversion of a receiver input signal into an acoustic signal for transmission towards the other one of the eardrums of the user, and wherein the receiver input of one of the first and second receivers is connected to a signal
  • a new method is provided of binaural signal enhancement in a binaural hearing aid system, the method comprising the steps of providing at least one microphone audio signal in response to sound, and providing an estimate of one of a target signal and a noise signal based on the at least one audio signal, phase shifting the estimate of one of the target signal and the noise signal, and providing a phase shifted signal representing the at least one microphone audio signal in which the phase shifted estimate of one of the target signal and the noise signal has substantially substituted the respective original one of the target signal and the noise signal, and transmitting a signal representing the phase shifted signal towards one of the eardrums of a user of the binaural hearing aid system, and transmitting a signal representing the at least one microphone audio signal towards the other one of the eardrums of the user.
  • the phase shifted estimate can exactly substitute the respective original signal; however typically, the estimate of a signal will deviate from the original signal and substitution of the original signal with its estimate will typically not lead to substitution of the deviation, and thus the estimate is said to substantially substitute the original signal.
  • one signal is said to represent another signal when the one signal is a function of the other signal, for example the one signal may be formed by analogue-to-digital conversion, or digital-to analogue conversion of the other signal; or, the one signal may be formed by conversion from another acoustic signal to an electronic signal or vice versa; or the one signal may be formed by analogue or digital filtering or mixing of the other signal; or the one signal may be formed by transformation, such as frequency transformation, etc, of the other signal; etc.
  • signals that are processed by specific circuitry may be identified by a name that may be used to identify any analogue or digital signal forming part of the signal path from the source of the signal in question to an input of the circuitry, e.g. signal processor, in question.
  • a name e.g. a name that may be used to identify any analogue or digital signal forming part of the signal path from the source of the signal in question to an input of the circuitry, e.g. signal processor, in question.
  • an output signal of a microphone i.e. the microphone audio signal
  • the at least one microphone may contain a single microphone; however preferably, the at least one microphone has two microphones. Further, the at least one microphone may have more than two microphones for improved separation of the target signal and the noise signal.
  • the second hearing aid may also comprise at least one microphone for provision of microphone audio signals in response to sound received at the respective microphones.
  • the transceiver of the first hearing aid is connected for reception of signals representing the microphone audio signals of the second hearing aid, and the signal separation unit is configured to provide the estimate of the target signal and the estimate of the noise signal based on the audio signals of the first and second hearing aids.
  • the phase shift circuit phase shifts the estimate of the target signal, and preferably, the phase shift ranges from 150° to 210°, more preferred the phase shift is approximately equal to 180°, and most preferred equal to 180°.
  • the signal separation unit may be configured to provide the estimates based on spectral characteristics of the audio signals as is well-known in the art of noise reduction.
  • the noise estimate is not suppressed in the output presented to the user; rather the target estimate and the noise estimate is presented to the user in a way that improves the user's SRT.
  • the signal separation unit may be configured to provide the estimates based on statistical characteristics of the audio signals as is well-known in the art of noise reduction.
  • the noise estimate is not suppressed in the output presented to the user; rather the target estimate and the noise estimate is presented to the user in a way that improves the user's SRT.
  • the signal separation unit may comprise a beamformer, and the beam former may be configured to provide the estimates based on microphone audio signals of the first and second hearing aids.
  • the beamformer of the signal separation unit is different from conventional beamformers in that the noise estimate is not suppressed in the output presented to the user; rather the target estimate and the noise estimate is presented to the user in a way that improves the user's SRT.
  • the beamformer combines the microphone audio signals output by a plurality of microphones of the at least one microphone into a target signal with varying sensitivity to sound sources in different directions in relation to the plurality of microphones.
  • a plot of the varying sensitivity as a function of the direction is denoted the directivity pattern.
  • a directivity pattern has at least one direction wherein the microphone signals substantially cancel each other.
  • such a direction is denoted a null direction.
  • a directivity pattern may comprise several null directions depending on the number of microphones in the plurality of microphones and depending on the signal processing.
  • the beamformer may be a fixed beamformer with a directional pattern that is fixed with relation to the head of the user.
  • the beamformer may for example be based on at least two microphones, with a directional pattern that has a maximum in the front direction of the user, i.e. the forward looking direction of the user, and a null in the opposite direction, i.e. the rear direction of the user.
  • the beamformer may be based on more than two microphones, and may include microphones of both hearing aids using wireless or wired communication techniques.
  • the increased distance between the microphones may be utilized to form a directional pattern with a narrow beam providing improved spatial separation of the target estimate from the noise estimate.
  • the conventional output of the beamformer may be used as the target estimate, and the noise estimate may be provided by subtraction of the target estimate from the microphone audio signal of one of the microphones of the plurality of microphones.
  • the respective microphone signals When microphones of both hearing aids of the binaural hearing aid system cooperate with the beamformer, the respective microphone signals must be sampled substantially synchronously. Time shifts as small as 20 -30 ⁇ S between sampling instants of the respective microphone signals in the two hearing aids may lead to a perceivable shift in the beam direction. Furthermore, a slowly time varying time shift between the sampling instants of the respective microphone signals, which inevitably will occur if the hearing aids are operated asynchronously, will result in an acoustic beam that appears to drift and focus in alternating directions.
  • the hearing aids of the binaural hearing aid system may be synchronized as for example discloses in more detail in WO 02/07479 .
  • the beamformer may comprise adaptive filters configured to filter respective microphone audio signals and to adapt the respective filter coefficients for adaptive beamforming towards a sound source. For example, the beamformer may adapt to optimize the signal to noise ratio.
  • An adaptable beamformer makes it possible to focus on a moving sound source or to focus on a non-moving sound source, while the user of the hearing aid system is moving. Furthermore, the adaptable beamformer is capable of adapting to changes in the sound environment, such as appearance of a new sound source, disappearance of a noise source or movement of noise sources relative to the user of the hearing aid system.
  • the noise can consist of both directional noise and other types of noise such as diffuse noise or babble noise.
  • Finding a solution to this optimization could be done adaptively using least mean square, recursive least square, steepest descent or other types of numerical optimization algorithms.
  • the signals are presented to the user in such a way that the SRT of the user is improved.
  • the target estimate is presented in opposite phase, i.e. 180° phase shifted with relation to each other, at the two ears of the user, while the noise estimate is presented in phase at the two ears of the user.
  • a first adder may be connected to the signal separation unit, and output a sum of the target estimate and the noise estimate provided by the signal separation unit, and the output of the first adder may be connected to a signal processor for further processing, such as hearing loss compensation, and the output of the signal processor may be connected to an output transducer that outputs a corresponding output to one ear of the user, or the output of the first adder may be connected directly to the output transducer.
  • a second adder may be connected to the signal separation unit, and output a sum of the reverse phases target estimate and the noise estimate provided by the signal separation unit, and the output of the second adder is connected to a transceiver that transmits the output of the second adder to the other hearing aid having a transceiver for reception of the output of the second adder.
  • the output of the transceiver may be connected to a signal processor for further processing, such as hearing loss compensation, and the output of the signal processor may be connected to an output transducer that outputs a corresponding output to another ear of the user, or the output of the transceiver may be connected directly to the output transducer.
  • the noise signal may be presented in opposite phase, i.e. 180° phase shifted with relation to each other, at the two ears of the user, while the target estimate is presented in phase at the two ears of the user.
  • the first hearing aid includes a delay between the adder and the output transducer so that the relative phase of the signals output by the respective output transducers of the first and second hearing aids is maintained.
  • the improvement of SRT as a function of the phase shift has a maximum at 180°; however the function is sine-shape with a flat maximum so that the improvement obtained by a phase shift ranging from 150° to 210° is close to the maximum improvement.
  • the phase shift need not be exactly 180°, but preferably has a value within the range from 135° to 225°, more preferred from 150° to 210°.
  • the new binaural hearing aid system may comprise a multi-channel first hearing aid in which the microphone audio signals are divided into a plurality of frequency channels.
  • individual target signal estimates and noise estimates may be provided in each frequency channel of the plurality of frequency channels, or may be provided in one or more selected frequency channels of the plurality of frequency channels, or one or more target signal estimates and noise estimates may be provided for one or more respective groups of selected frequency channels of the plurality of frequency channels, or one target signal estimate and noise estimate may be provided based on all the frequency channels of the plurality of frequency channels.
  • the plurality of frequency channels may include warped frequency channels, for example all of the frequency channels may be warped frequency channels.
  • the new binaural hearing aid system may additionally provide circuitry used in accordance with other conventional methods of hearing loss compensation so that the new circuitry or other conventional circuitry can be selected for operation as appropriate in different types of sound environment.
  • the different sound environments may include speech, babble speech, restaurant clatter, music, traffic noise, etc.
  • the new binaural hearing aid system may for example comprise a Digital Signal Processor (DSP), the processing of which is controlled by selectable signal processing algorithms, each of which having various parameters for adjustment of the actual signal processing performed.
  • DSP Digital Signal Processor
  • the gains in each of the frequency channels of a multi-channel hearing aid are examples of such parameters.
  • One of the selectable signal processing algorithms operates in accordance with the new method.
  • various algorithms may be provided for conventional noise suppression, i.e. attenuation of undesired signals and amplification of desired signals.
  • Microphone audio signals obtained from different sound environments may possess very different characteristics, e.g. average and maximum sound pressure levels (SPLs) and/or frequency content. Therefore, each type of sound environment may be associated with a particular program wherein a particular setting of algorithm parameters of a signal processing algorithm provides processed sound of optimum signal quality in a specific sound environment.
  • a set of such parameters may typically include parameters related to broadband gain, corner frequencies or slopes of frequency-selective filter algorithms and parameters controlling e.g. knee-points and compression ratios of Automatic Gain Control (AGC) algorithms.
  • AGC Automatic Gain Control
  • Signal processing characteristics of each of the algorithms may be determined during an initial fitting session in a dispenser's office and programmed into the new binaural hearing aid system in a non-volatile memory area.
  • the new binaural hearing aid system may have a user interface, e.g. buttons, toggle switches, etc, of the hearing aid housings, or a remote control, so that the user of the new binaural hearing aid system can select one of the available signal processing algorithms to obtain the desired hearing loss compensation in the sound environment in question.
  • a user interface e.g. buttons, toggle switches, etc
  • the user of the new binaural hearing aid system can select one of the available signal processing algorithms to obtain the desired hearing loss compensation in the sound environment in question.
  • the new binaural hearing aid system may be capable of automatically classifying the user's sound environment into one of a number of sound environment categories, such as speech, babble speech, restaurant clatter, music, traffic noise, etc, and may automatically select the appropriate signal processing algorithm accordingly as known in the art.
  • Fig. 1 schematically illustrates an example of the new binaural hearing aid system 10.
  • the new binaural hearing aid system 10 has first and second hearing aids 10A, 10B.
  • the second hearing aid 10B has a receiver 48B and a transceiver (not shown) for reception of the input signal to the receiver 48B from the first hearing aid 10A by wired or wireless transmission.
  • the acoustic output signal emitted by the second hearing aid 10B is controlled by the first hearing aid 10A.
  • the first hearing aid 10A comprises one microphone 14 for provision of microphone audio signal 18 in response to sound received at the microphone 14.
  • the microphone audio signal 18 may be pre-filtered in respective pre-filters (not shown) well-known in the art, and input to the signal separation unit 12.
  • the signal separation unit 12 estimates the target signal and subtracts two times the estimated target signal from the microphone audio signal 18 to obtain a signal, in the following denoted "the phase shifted signal", representing the microphone audio signal 18; however, wherein the original target signal has been replaced by the estimate of the target signal phase shifted by 180°.
  • the phase shifted signal is output to a transceiver (not shown) in the first hearing aid 10A for transmission to the second hearing aid 10B.
  • a receiver 48 of the first hearing aid 10A converts the microphone audio signal 18 into an acoustic signal for transmission towards the eardrum of one ear of the user
  • the receiver 48B of the second hearing aid 10B converts the phase shifted signal into an acoustic signal for transmission towards the eardrum of the other ear of the user thereby improving BMLD and SRT.
  • the signal separation unit 12 may be configured to provide the estimate based on time-domain, spectral, and/or statistical characteristics of the microphone audio signal as is well-known in the art of noise reduction.
  • further processing may be applied to the respective signals before input to the respective receivers 48, 48B, e.g. for hearing loss compensation of the respective signals.
  • the new binaural hearing aid system (10) shown in Fig. 2 is similar to the hearing aid system shown in Fig. 1 except for the fact that the signal separation unit 12 shown in Fig. 2 is configured to provide both an estimate of the target signal 26 and an estimate of the noise signal 30 based on the possibly pre-filtered microphone audio signal 18.
  • the estimate of the target signal 26 is added to the estimate of the noise signal 30 in a first adder 42 and the output sum of the estimate of the target signal 26 and the estimate of the noise signal 30 is input to an output transducer 48 that converts the output of first adder 42 into an acoustic output signal that is transmitted towards the eardrum of the user wearing the binaural hearing aid system 10.
  • the estimate of the target signal 26 is subtracted; corresponding to a phase shift of 180°, from the estimate of the noise signal 30 in a second adder 50, and the output of the second adder 50 is transmitted output transducer 48Bfor conversion into an acoustic output signal that is transmitted towards the other eardrum of the user wearing the binaural hearing aid system 10.
  • the BMLD and SRT are improved.
  • the estimate of the target signal 26 and the estimate of the noise signal 30 may be swapped so that the estimate of the noise signal 20 is phase shifted 180° before presentation to one of the eardrums of the user instead of phase shifting the estimate of the target signal 26.
  • the improvement in BMLD and SRT obtained in this way is smaller than the improvement obtained by phase shift of the estimate of the target signal 26.
  • the new binaural hearing aid system (10) shown in Fig. 3 is similar to the hearing aid system shown in Fig. 1 except for the fact that a microphone audio signal 18B output by a microphone 14B in the second hearing aid 10B is transmitted by wired or wireless transmission to the first hearing aid 10A and input to the signal separation unit 12 so that the signal separation unit 12 can base the estimate of the target signal on both microphone audio signals 18, 18B, e.g. by beamforming as explained further below.
  • the new binaural hearing aid system (10) shown in Fig. 4 is similar to the hearing aid system shown in Fig. 3 except for the fact that the signal separation unit 12 shown in Fig. 4 , like the signal separation unit shown in Fig. 2 , is configured to provide both an estimate of the target signal 26 and an estimate of the noise signal 30 based on the possibly pre-filtered microphone audio signal 18.
  • the estimate of the target signal 26 is added to the estimate of the noise signal 30 in a first adder 42 and the output sum of the estimate of the target signal 26 and the estimate of the noise signal 30 is input to an output transducer 48 that converts the output of first adder 42 into an acoustic output signal that is transmitted towards the eardrum of the user wearing the binaural hearing aid system 10.
  • the estimate of the target signal 26 is subtracted; corresponding to a phase shift of 180°, from the estimate of the noise signal 30 in a second adder 50, and the output of the second adder 50 is transmitted output transducer 48Bfor conversion into an acoustic output signal that is transmitted towards the other eardrum of the user wearing the binaural hearing aid system 10.
  • the BMLD and SRT are improved.
  • Fig. 5 schematically illustrates a digital signal separation unit 12 including an adaptive beamformer 10 with two microphones 14, 16.
  • the microphone audio signals 18, 20 are pre-filtered in conventional pre-filters 22, 24 before beamforming.
  • the microphone audio signals 18, 20 may be digitized before or after the pre-filters 22, 24 by A/D converters (not shown). Signals before and after prefiltering and before and after analogue-digital conversion are all termed microphone audio signals.
  • the output 26 of first subtractor 28 generates the estimate of the target signal from the assumed target direction using adaptive beamforming.
  • the estimate of the target signal 26 is subsequently presented to one of the two ears of the user and in opposite phase to the other of the two ears of the user.
  • the output 30 of the adaptive filter 32 filtering the output of second subtractor 34 generates the noise estimate for subsequent presentation to both ears of the user.
  • h 2 ( n ) is the impulse response of sound propagation from the source emitting the signal s(n) to the second microphone 16
  • g 2 ( n ) is the the impulse response of sound propagation from the noise source emitting the signal q ( n ) to the second microphone 16.
  • the output 26 of the target signal is equal to h 1 ( n ) ⁇ s ( n ), and the output 30 of the noise estimate is equal to g 1 ( n ) ⁇ q ( n ).
  • Fig. 6 schematically illustrates a signal separation unit 12 based on four microphones 22, 24, 22B, 24B, two of which 22, 24 are located in the first hearing aid 10A and other two of which 22B, 24B are located in the second hearing aid 10B.
  • the increased distance between the microphones may be utilized to form a directional pattern with a narrow beam providing improved spatial separation of the target estimate from the noise estimate.
  • the conventional output of the beamformer may be used as the target estimate, and the noise estimate may be provided by subtraction of the target estimate from the microphone audio signal of one of the microphones in the plurality of microphones.
  • the microphone audio signals 18, 20 of the two microphones 22, 24 of the first hearing aid 10 are pre-filtered in respective pre-filters 22, 24 well-known in the art, into microphone audio signals y 1 (n), y 2 (n) and input to respective adaptive filters a 1 (n), a 2 (n).
  • the pre-filtered microphone audio signals of the two microphones 22B, 24B of the second hearing aid 10B are encoded for transmission in the second hearing aid 10B and transmitted to the first hearing aid 10A using wireless or wired data transmission.
  • the transmitted data representing the microphone audio signals of the two microphones 22B, 24B of the second hearing aid 10B are received by the transceiver 36 of the first hearing aid 10A and decoded in decoder 38 into two microphone audio signals y 3 (n), y 4 (n) and input to respective adaptive filters a 3 (n), a 4 (n).
  • the adaptive filters a 1 (n), a 2 (n), a 3 (n), a 4 (n) are configured to filter the respective microphone audio signals y 1 (n), y 2 (n), y 3 (n), y 4 (n) and to adapt the respective filter coefficients for adaptive beamforming towards a sound source.
  • the adaptable filters a 1 (n), a 2 (n), a 3 (n), a 4 (n) make it possible to focus on a moving sound source or to focus on a non-moving sound source, while the user of the hearing aid system is moving. Furthermore, the adaptable filters a 1 (n), a 2 (n), a 3 (n), a 4 (n) are capable of adapting to changes in the sound environment, such as appearance of a new sound source, disappearance of a noise source or movement of noise sources relative to the user of the hearing aid system.
  • the noise can consist of both directional noise and other types of noise such as diffuse noise or babble noise.
  • Filter adaptation is preferably performed using the least mean square (LMS) algorithm, more preferred the normalized least means square (NLMS) algorithm; however other algorithms may also be used, such as recursive least square, steepest descent or other types of numerical optimization algorithms.
  • LMS least mean square
  • NLMS normalized least means square
  • the signals are presented to the user in such a way that the SRT of the user is improved as schematically illustrated in Fig. 7 .
  • Fig. 7 shows an example of the new binaural hearing aid system 10.
  • the new binaural hearing aid system 10 has first and second hearing aids 10A, 10B with transceivers 36, 36B for data communication between the two hearing aids 10A, 10B.
  • the first hearing aid 10A comprises at least one microphone with two microphones 14, 16 for provision of microphone audio signals 18, 20 in response to sound received at the respective microphones 14, 16.
  • the microphone audio signals 18, 20 are pre-filtered in respective pre-filters 22, 24 well-known in the art, into microphone audio signals and input to the signal separation unit 12.
  • the signal separation unit 12 is shown in more detail in Fig. 6 and explained above with reference to Fig. 6 .
  • the second hearing aid 10B also comprises at least one microphone with two microphones 14B, 16B for provision of microphone audio signals 18B, 20B in response to sound received at the respective microphones 14B, 16B.
  • the microphone audio signals 18B, 20B are pre-filtered by pre-filters 22B, 24B as is well-known in the art.
  • the pre-filtered microphone audio signals of the two microphones 22B, 24B are encoded in Codec 40B for transmission to the first hearing aid 10A using wireless data transmission.
  • the transmitted data representing the microphone audio signals of the second hearing aid 10B are received by the transceiver 36 of the first hearing aid 10A and decoded in decoder 38 into two microphone audio signals that are input to the signal separation unit 12 as explained above with reference to Fig. 6 .
  • the signal separation unit 12 is configured to provide the estimate of the target signal 26 and the estimate of the noise signal 30 based on the pre-filtered microphone audio signals of the first and second hearing aids 10A, 10B.
  • the conventional output of the beamformer is used as the estimate of the target signal 26, and the estimate of the noise signal 30 is provided by subtraction of the estimate of the target signal 26 from the pre-filtered microphone audio signal of one of the microphones in the plurality of four microphones 14,16, 14B, 16B.
  • the signals are presented to the user in such a way that the SRT of the user is improved:
  • the estimate of the target signal 26 is added to the estimate of the noise signal 30 in a first adder 42 and the output sum of the estimate of the target signal 26 and the estimate of the noise signal 30 is delayed in delay 44 and input to a signal processor 46 for hearing loss compensation.
  • the delay 44 maintains the desired relative phase of the signals output by the first and second hearing aids 10A, 10B, respectively.
  • An output transducer 48 in the illustrated example a receiver 48, converts the output of the signal processor 46 into an acoustic output signal that is transmitted towards the eardrum of the user wearing the binaural hearing aid system 10.
  • the estimate of the target signal 26 is subtracted; corresponding to a phase shift of 180°, from the estimate of the noise signal 30 in a second adder 50, and the output of the second adder 50 is encoded in Codec 40 for transmission by transceiver 36 to the second hearing aid 10B.
  • the transmitted sum is received by the transceiver 36B and decoded by decoder 38B and input to signal processor 46B for hearing loss compensation.
  • An output transducer 48B in the illustrated example a receiver 48B, converts the output of the signal processor 46B into an acoustic output signal that is transmitted towards the eardrum of the user wearing the binaural hearing aid system 10. In this way, the SRT of the user may be improved up to 20 dB depending on the sound environment.
  • the estimate of the target signal 26 and the estimate of the noise signal 30 may be swapped so that the estimate of the noise signal 20 is phase shifted 180° before presentation to one of the eardrums of the user instead of phase shifting the estimate of the target signal 26.
  • the improvement in SRT obtained in this way is smaller than the improvement obtained by phase shift of the estimate of the target signal 26.

Claims (7)

  1. Système de prothèse auditive binaurale (10) comprenant
    une pluralité de microphones (14, 16, 14B, 16B) pour fournir une pluralité respective de signaux audio de microphone (18, 20, 18B, 20B) en réponse à un son reçu au niveau de la pluralité de microphones (14, 16, 14B, 16B),
    une unité de séparation de signal avec un formeur de faisceaux (12) configurée pour fournir une estimation d'un signal cible (26) et une estimation d'un signal de bruit (30) d'après la pluralité de signaux audio de microphone (18, 20, 18B, 20B), dans lequel le formeur de faisceaux (12) comprend des filtres adaptatifs (32) dont chacun est configuré pour filtrer un signal respectif de la pluralité de signaux audio de microphone (18, 20, 18B, 20B),
    caractérisé en ce que
    le formeur de faisceaux (12) comporte un additionneur (34) avec une pluralité d'entrées dont chacune est connectée à une sortie d'un filtre respectif des filtres adaptatifs (32) et
    une sortie pour fournir la somme des sorties des filtres adaptatifs (32), et
    dans lequel
    la sortie est connectée à une entrée de chacun des filtres adaptatifs (32) pour fournir la somme des sorties des filtres adaptatifs (32) en tant que fonction d'erreur à chacun des filtres adaptatifs (32), moyennant quoi chacun des filtres adaptatifs (32) est configuré pour adapter les coefficients de filtre pour minimiser la somme des sorties des filtres adaptatifs (32) pour fournir l'estimation du signal cible (26), et
    dans lequel le formeur de faisceaux (12) comprend un soustracteur (28) configuré pour soustraire l'estimation du signal cible (26) à partir de l'un de la pluralité de signaux audio de microphone (18, 20, 18B, 20B) pour fournir l'estimation du signal de bruit (30).
  2. Système de prothèse auditive binaurale (10) selon la revendication 1, dans lequel les filtres adaptatifs (32) sont configurés pour déterminer de façon adaptative les coefficients de filtre en résolvant le problème d'optimisation : a i n i = 1 N = arg min a i n i = 1 N ‖ z n ‖ 2
    Figure imgb0016
     sous réserve de ∑ i = 1 N a i n ∗ h i n = h 1 n
    Figure imgb0017
     dans lequel
    z(n) est la somme des sorties des filtres adaptatifs (32),
    n est le nombre d'échantillons,
    N est le nombre de filtres adaptatifs (32),
    i est le nombre-indice des filtres adaptatifs (32),
    ai(n) est le nième coefficient de filtre du iième filtre adaptatif, et
    hi(n) est le nième échantillon de la réponse impulsionnelle de propagation de son depuis la source émettant le signal au iième microphone de la pluralité de microphones (14, 16, 14B, 16B).
  3. Système de prothèse auditive binaurale (10) selon l'une quelconque des revendications précédentes, dans lequel le nombre de filtres adaptatifs est de 4.
  4. Système de prothèse auditive binaurale (10) selon l'une quelconque des revendications précédentes, comprenant en outre
    une première prothèse auditive (10A) comprenant au moins un microphone (14, 16) pour fournir au moins un signal audio de microphone (18, 20) respectif en réponse à un son reçu au niveau de le au moins un microphone (14, 16), et
    une seconde prothèse auditive (10B) comprenant au moins un microphone (14B, 16B) pour fournir au moins un signal audio de microphone (18B, 20B) respectif en réponse à un son reçu au niveau de le au moins un microphone (14B, 16B), et dans lequel
    un émetteur-récepteur (36B) dans la seconde prothèse auditive (10B) est connecté pour transmettre des signaux représentant le au moins un signal audio de microphone (18B, 20B) à la première prothèse auditive (10A), et dans lequel
    un émetteur-récepteur (36) dans la première prothèse auditive (10A) est connecté pour recevoir les signaux représentant le au moins un signal audio de microphone (18B, 20B) de la seconde prothèse auditive (10B), et dans lequel
    l'unité de séparation de signal (12) est configurée pour fournir l'estimation de l'un du signal cible (26) et du signal de bruit (30) d'après le au moins un signal audio de microphone (18, 20, 18B, 20B) des première et seconde prothèses auditives (10A, 10B).
  5. Système de prothèse auditive binaurale (10) selon l'une quelconque des revendications précédentes, comprenant en outre
    un circuit de déphasage configuré pour déphaser l'estimation de l'un du signal cible (26) et du signal de bruit (30), et
    un additionneur de déphasage (50) connecté pour fournir un signal déphasé représentant un son reçu au niveau de la pluralité de microphones (14, 16, 14B, 16B) dans lequel le déphasage de l'estimation de l'un du signal cible (26) et du signal de bruit (30) a sensiblement remplacé le signal d'origine respectif du signal cible (26) et du signal de bruit (30), et
    un premier récepteur (48) pour convertir un signal d'entrée de récepteur en un signal acoustique pour le transmettre vers l'un des tympans d'un utilisateur du système de prothèse auditive binaurale (10), et
    un second récepteur (48B) pour convertir un signal d'entrée de récepteur en un signal acoustique pour le transmettre vers l'autre des tympans de l'utilisateur, et dans lequel l'entrée de récepteur de l'un des premier et second récepteurs (48, 48B) est connectée à un signal représentant le signal déphasé, et
    l'entrée de récepteur de l'autre des premier et second récepteurs (48B, 48) est connectée à un signal représentant la somme de l'estimation du signal cible (26) et de l'estimation du signal de bruit (30).
  6. Système de prothèse auditive binaurale (10) selon la revendication 5, dans lequel le circuit de déphasage déphase l'estimation du signal cible (26).
  7. Système de prothèse auditive binaurale (10) selon la revendication 5 ou 6, dans lequel le déphasage varie de 150° à 210°.
EP16164585.8A 2011-12-30 2011-12-30 Prothèse auditive avec amélioration de signal Revoked EP3059979B1 (fr)

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EP2611215B1 (fr) 2016-04-20
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DK3059979T3 (da) 2020-06-08
EP2611215A1 (fr) 2013-07-03

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