EP2683179B1 - Hörgerät mit Frequenzdemaskierung - Google Patents

Hörgerät mit Frequenzdemaskierung Download PDF

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Publication number
EP2683179B1
EP2683179B1 EP20120175247 EP12175247A EP2683179B1 EP 2683179 B1 EP2683179 B1 EP 2683179B1 EP 20120175247 EP20120175247 EP 20120175247 EP 12175247 A EP12175247 A EP 12175247A EP 2683179 B1 EP2683179 B1 EP 2683179B1
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EP
European Patent Office
Prior art keywords
signal
estimate
hearing aid
masker
frequency
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EP20120175247
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English (en)
French (fr)
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EP2683179A1 (de
Inventor
Karl-Fredrik Johan Gran
Andrew Burke Dittberner
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GN Hearing AS
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GN Resound AS
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Priority to DK12175247.1T priority Critical patent/DK2683179T3/en
Priority to EP20120175247 priority patent/EP2683179B1/de
Priority to US13/547,720 priority patent/US9185499B2/en
Priority to JP2012279621A priority patent/JP2013153427A/ja
Priority to CN201210593890.2A priority patent/CN103546849B/zh
Publication of EP2683179A1 publication Critical patent/EP2683179A1/de
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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
    • H04R25/35—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception 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; 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
    • H04R25/43—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; 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
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/03—Synergistic effects of band splitting and sub-band processing
    • 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 hearing aid is provided 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.
  • US 2010/0067721 shows a hearing device where a frequency transposition is applied only to useful signals.
  • a new and inventive method of enhancement of a target signal is disclosed.
  • the new method makes use of the human auditory system's capability of concentrating on a target signal.
  • a new and inventive hearing aid using the new method is also disclosed.
  • a target signal is a signal representing sound that a person desires to listen to.
  • Speech, music, and sounds from the nature are examples of target signals.
  • a masker signal is also a signal representing sound; however, sound that a person perceives to be interfering with the target signal in an undesirable way, e.g. making it difficult for the person to understand or enjoy the target signal.
  • the masker signal can be background speech, restaurant clatter, music (when speech is the target signal), traffic noise, etc.
  • 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.
  • speech intelligibility is increased by outputting the target signal and the masker signal to the eardrums of a user of the binaural hearing aid in different frequency bands.
  • the improvement is obtained without removal a part of the signal; rather, the target and masker signals are presented to the eardrums of the user in a way that the auditory system of the user's auditory system can perform natural noise reduction and separate the target signal from the masker signal.
  • a new hearing aid comprising at least one microphone for provision of 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 a target signal and a masker signal based on the at least one microphone audio signal, a frequency modifying unit configured to modify the frequency content of at least one of the estimates of the target signal and the masker signal so that, upon processing by the frequency modifying unit, the estimated target signal and the estimated masker signal are output substantially in different frequency bands, a receiver for conversion of a combination of the estimate of the target signal and the estimate of the masker signal as modified and output by the frequency modifying unit into an acoustic signal for transmission towards one of the eardrums of a user of the binaural hearing aid system.
  • two signals are said to be output substantially in different frequency bands of the hearing aid, when in a majority of the frequency bands, e.g. in more than 51 % of the frequency bands of the hearing aid, and in a major part of the time, e.g. in more than 51 % of the time; one of the signals, e.g. the target signal, in one of the frequency bands, and the other one of the signals, e.g. the masker signal, in another one of the frequency bands, have a signal level that is less than 20 %, such as less than 15 %, preferably less than 10 %, more preferred less than 5 %, most preferred less than 1 % of the signal level of the other signal.
  • the signal level may be an RMS-value, a peak value, an average amplitude value, etc, as defined in a predetermined time period.
  • the new hearing aid may constitute a first hearing aid of a new binaural hearing aid system further having a second hearing aid 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, and wherein a transceiver in the second hearing aid is connected for transmission of signals representing the at least one microphone audio signal to the first hearing aid, and wherein a transceiver in the first hearing aid is connected for reception of the signals representing the at least one microphone audio signal of the second hearing aid, and wherein the signal separation unit is configured to provide the estimate of the target signal and the estimate of the masker signal based on the at least one microphone audio signals of the first and second hearing aids.
  • the frequency modifying unit may be configured to frequency shift one of the estimate of the target signal and the estimate of the masker signal into a frequency region where the other one of the estimate of the target signal and the estimate of the masker signal substantially is not present.
  • the frequency modifying unit may be configured to determine pitch and harmonics of the estimate of the target signal and the estimate of the masker signal, respectively, and frequency shift one of the estimate of the target signal and the estimate of the masker signal so that pitch and harmonics of the frequency shifted signal resides in between respective pitch and harmonics of the other signal.
  • the frequency modifying unit may comprise a filter-bank, and the frequency modifying unit may be configured to filter the estimate of the target signal with frequency bands assigned to the estimate of the target signal, and filter the estimate of the masker signal with other frequency bands assigned to the estimate of the masker signal.
  • the filter-bank may be tuned to the auditory filters of the intended user.
  • the frequency modifying unit may also be configured to phase shift the estimate of the target signal with relation to the estimate of the masker signal.
  • the target signal is designated S and the masker signal is designated N
  • the incoming sound signal is S+N.
  • the target signal S is estimated, and the estimate is denoted ES.
  • An estimate of the masker signal N is designated EN and may be determined by subtracting the estimate ES from the sound signal S+N.
  • the estimate of the target signal ES may be phase shifted 180° with relation to the estimate of the masker signal EN by subtracting two times ES from the incoming signal S+N with the result: S+N-ES-ES. Since ES is approximately equal to S, the result of the subtraction is approximately: N-ES which is approximately equal to -ES + EN, i.e. the estimate of the target signal ES has been phase shifted substantially by 180° with relation to the estimate of the masker signal. This operation may be performed before or after frequency modification in the frequency modifying unit.
  • phase shift of exactly or approximately 180 ° is not required in order to obtain a significant improvement of SRT; rather such improvement is obtained with phase shift in the range from 135 ° - 225 ° , such as from 150° - 210°.
  • 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 as disclosed above.
  • both the target signal S and the masker signal 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 as disclosed above.
  • This operation may be performed before or after frequency modification in the frequency modifying unit.
  • the target signal S and the masker signal may be swapped so that the masker signal estimate is phase shifted instead of the target signal for improved BMLD and SRT as disclosed above; however with decreased performance compared to phase shifting the target signal S.
  • 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 consist of a single microphone; however preferably, the at least one microphone comprises a single microphone, e.g. a plurality of microphones, such as two microphones.
  • the at least one microphone may have more than two microphones for improved separation of the target signal and the masker 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 masker signal based on the audio signals of the first and second hearing aids.
  • the frequency modifying unit phase shifts the estimate of the target signal with relation to the estimate of the masker signal with a phase shift ranging from 150° to 210°, more preferred the phase shift is approximately equal to 180°, and most preferred equal to 180°.
  • 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 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 masker signal estimate is presented in phase at the two ears of the user.
  • 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 masker signal estimate is not suppressed in the output presented to the user; rather the target estimate and the masker signal estimate is presented to the user in a way that improves the user's SRT as disclosed above.
  • 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 masker signal estimate is not suppressed in the output presented to the user; rather the target estimate and the masker signal 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 masker signal estimate is not suppressed in the output presented to the user; rather the target estimate and the masker signal estimate is presented to the user in a way that improves the user's SRT.
  • the beamformer combines the microphone audio signals output by the plurality of microphones 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 of 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 masker signal estimate.
  • the conventional output of the beamformer may be used as the target estimate, and the masker signal 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 masker signal or noise source or movement of masker signal or noise sources relative to the user of the hearing aid system.
  • the masker signal 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 signal separation unit is configured in such a way that the estimate of the target signal and the estimate of the masker signal include the spatial cues of the original signal. This can be achieved by appropriate microphone placements and/or proper pre/post processing of the microphone signals.
  • Each of the hearing aids of the binaural hearing aid may have a signal separation unit so that an estimate of the target signal and an estimate of the masker signal are available in each of the hearing aids, preferably with correct spatial cues.
  • the signals are presented to the user in such a way that the SRT of the user is improved as disclosed above.
  • 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 masker signal 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 masker signal 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 masker signal 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 or noise signals and amplification of target 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 10 that operates to enhance a target signal making use of the human auditory system's capability of concentrating on a target signal.
  • speech intelligibility is increased by outputting a target signal and a masker signal to the eardrums of a user of the binaural hearing aid in different frequency bands.
  • the improvement is obtained without removal of a part of the signal; rather, the target and masker signals are presented to the eardrums of the user in a way that the auditory system of the user's auditory system can perform natural noise reduction and separate the target signal from the masker signal.
  • the illustrated new hearing aid 10 comprises a microphone 14 for provision of a 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 is configured to provide an estimate of a target signal 26 and a masker signal 30 based on the microphone audio signal 18, a frequency modifying unit 52 configured to modify the frequency content of at least one of the estimates of the target signal 26 and the masker signal 30 so that, upon processing by the frequency modifying unit 52, the estimated target signal 26 and the estimated masker signal 30 are output substantially in different frequency bands.
  • a hearing loss processor 46 configured for hearing loss compensation as is well-known in the art of hearing aids processes a combination 32 of the estimated target signal 26 and estimated masker signal 30 as modified by the frequency modifying unit 52 into a hearing loss compensated audio signal 34, and an output transducer 48, in the illustrated example a receiver 48, converts the output 34 of the hearing loss processor 46 into an acoustic output signal that is transmitted towards the eardrum of the user wearing the hearing aid 10.
  • the frequency modifying unit 52 may be configured to frequency shift one of the estimate of the target signal 26 and the estimate of the masker signal 30 into a frequency region where the other one of the estimate of the target signal 26 and the estimate of the masker signal 30 substantially is not present as illustrated in Fig. 2 showing a frequency shifted estimated masker signal 30.
  • the frequency modifying unit 52 may be configured to determine pitch and harmonics of the estimate of the target signal 26 and the estimate of the masker signal 30, respectively, and frequency shift one of the estimate of the target signal 26 and the estimate of the masker signal 30 so that pitch and harmonics of the frequency shifted signal resides in between respective pitch and harmonics of the other signal as illustrated in Fig. 3 .
  • the frequency modifying unit 52 may comprise a filter-bank as shown in Fig. 4 , and the frequency modifying unit 52 may be configured to filter the estimate of the target signal with frequency bands assigned to the estimate of the target signal 26, and filter the estimate of the masker signal 30 with other frequency bands assigned to the estimate of the masker signal 30.
  • the filter-bank may be tuned to the auditory filters of the intended user.
  • the masker signal can be background speech, restaurant clatter, music (when speech is the target signal), traffic noise, etc.
  • the microphone 14 may be substituted with two microphones, or an array of microphones with more than two microphones for improved separation of the target signal and the masker signal.
  • the signal separation unit 12 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 masker signal estimate is not suppressed in the output presented to the user; rather the target estimate and the masker signal estimate is presented to the user in a way that improves the user's SRT.
  • the signal separation unit 12 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 masker signal estimate is not suppressed in the output presented to the user; rather the target estimate and the masker signal estimate is presented to the user in a way that improves the user's SRT.
  • the signal separation unit 12 may comprise a beamformer.
  • the beamformer of the signal separation unit 12 is different from conventional beamformers in that the masker signal estimate is not suppressed in the output presented to the user; rather the target estimate and the masker signal estimate is presented to the user in a way that improves the user's SRT.
  • 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.
  • the conventional output of the beamformer may be used as the target estimate, and the masker signal 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 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 masker signal or noise source or movement of masker signal or noise sources relative to the user of the hearing aid system.
  • the signal separation unit is configured in such a way that the estimate of the target signal and the estimate of the masker signal include the spatial cues of the original signal. This can be achieved by appropriate microphone placements and/or proper pre/post processing of the microphone signals.
  • the signals are presented to the user in such a way that the SRT of the user is improved.
  • the new hearing aid 10 may be a multi-channel hearing aid in which the microphone audio signals are divided into a plurality of frequency channels.
  • individual target signal estimates and masker signal 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 masker signal 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 masker signal 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 hearing aid 10 may additionally provide circuitry 46 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 hearing aid 10 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 disclosed method of signal enhancement.
  • various algorithms may be provided for conventional noise suppression, i.e. attenuation of undesired or noise signals and amplification of target 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 hearing aid 10 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 hearing aid 10 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.
  • sound environment categories such as speech, babble speech, restaurant clatter, music, traffic noise, etc.
  • Fig. 5 shows a new binaural hearing aid system 10 with 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 provides an estimate of the target signal 26 and an estimate of the masker signal 30 based on the possibly pre-filtered microphone audio signal 18 and outputs the estimates to the frequency modifying unit 52.
  • the frequency modifying unit 52 modifies the frequency content of the estimates of the target signal 26 and the masker signal 30 so that the estimated target signal 26 and the estimated masker signal 30 are output substantially in different frequency bands, e.g. as illustrated in Figs. 2 - 4 , respectively.
  • the estimate of the target signal 26 is added to the estimate of the masker signal 30, at least one of which is frequency modified, in a first adder 42 and the output sum 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 masker signal 30, at least one of which is frequency modified, in a second adder 50, and the output of the second adder 50 is transmitted to output transducer 48B for 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 masker signal 30 may be swapped so that the estimate of the masker 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 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 as is well-known in the art of hearing aids.
  • the new binaural hearing aid system 10 shown in Fig. 6 is similar to the hearing aid system shown in Fig. 5 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 and the estimate of the masker signal 30 on both, possibly pre-filtered, microphone audio signals 18, 18B, e.g. by beamforming as explained further below.
  • the frequency modifying unit 52 modifies the frequency content of the estimates of the target signal 26 and the masker signal 30 so that the estimated target signal 26 and the estimated masker signal 30 are output substantially in different frequency bands, e.g. as illustrated in Figs. 2 - 4 , respectively.
  • the estimate of the target signal 26 is added to the estimate of the masker signal 30, at least one of which is frequency modified, in a first adder 42 and the output sum 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 frequency modified estimate of the target signal 26 is subtracted; corresponding to a phase shift of 180°, from the estimate of the masker signal 30, at least one of which is frequency modified, in a second adder 50, and the output of the second adder 50 is transmitted to output transducer 48B for 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 masker signal 30 may be swapped so that the estimate of the masker 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 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.
  • Fig. 7 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 masker signal 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 masker signal 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 )
  • the output 30 of the masker signal estimate is equal to g 1 ( n ) * q ( n ).
  • Fig. 8 schematically illustrates a signal separation unit 12 based on four microphones 14, 16, 14B, 16B, two of which 14, 16are located in the first hearing aid 10Aand other two of which (not shown) 14B, 16B are located in the second hearing aid 10B (not shown).
  • 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 masker signal estimate.
  • the conventional output of the beamformer may be used as the target estimate, and the masker signal 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 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 microphones14B, 16B (not shown) of the second hearing aid 10B (not shown) are encoded for transmission in the second hearing aid 10B (not shown) 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 14B, 16B (not shown)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 masker signal or noise source or movement of masker signal or noise sources relative to the user of the hearing aid system.
  • the masker signal can consist of both directional masker signal or noise and other types of masker signals or 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. 9 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. 8 and explained above with reference to Fig. 8 .
  • 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. 8 .
  • the signal separation unit 12 is configured to provide the estimate of the target signal 26 and the estimate of the masker 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 masker 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 of 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 masker signal 30, at least one of which is frequency modified, in a first adder 42 and the output sum of the estimate of the target signal 26 and the estimate of the masker 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 masker signal 30, at least one of which is frequency modified, 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 masker signal 30 may be swapped so that the estimate of the masker 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.

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Claims (16)

  1. Hörgerät (10), umfassend
    mindestens ein Mikrofon (14, 16) zum Bereitstellen mindestens eines Mikrofonaudiosignals (18, 20) als Reaktion auf Ton, der bei dem mindestens einen Mikrofon (14, 16) empfangen wird,
    eine Signaltrennungseinheit (12), die zum Bereitstellen einer Schätzung eines Zielsignals (26) und einer Schätzung eines Masker-Signals (30) auf der Basis des mindestens einen Mikrofonaudiosignals (18, 20) gestaltet ist,
    eine Frequenzmodifizierungseinheit (52), die zum Modifizieren des Frequenzgehalts mindestens einer der Schätzungen des Zielsignals (26) und des Masker-Signals (30) gestaltet ist,
    einen Empfänger (48) zum Umwandeln einer Kombination der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30), wie von der Frequenzmodifizierungseinheit (52) modifiziert und ausgegeben, in ein akustisches Signal zum Senden zu einem der Trommelfelle eines Benutzers des Hörgeräts (10),
    dadurch gekennzeichnet, dass
    die Frequenzmodifizierungseinheit (52) zum derartigen Modifizieren des Frequenzgehalts mindestens einer der Schätzungen des Zielsignals (26) und des Masker-Signals (30) gestaltet ist, dass bei Verarbeitung durch die Frequenzmodifizierungseinheit (52) das geschätzte Zielsignal (26) und das geschätzte Masker-Signal (30) im Wesentlichen in verschiedenen Frequenzbändern ausgegeben werden.
  2. Hörgerät (10) nach Anspruch 1, wobei die Frequenzmodifizierungseinheit (52) gestaltet ist zum
    Frequenzverschieben einer der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30) in einen Frequenzbereich, wo die andere der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30) im Wesentlichen nicht vorhanden ist.
  3. Hörgerät (10) nach Anspruch 1, wobei die Frequenzmodifizierungseinheit (52) gestaltet ist zum
    Bestimmen von Tonhöhe und Oberwellen der Schätzung des Zielsignals (26) bzw. der Schätzung des Masker-Signals (30) und
    Frequenzverschieben einer der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30), so dass Tonhöhe und Oberwellen des frequenzverschobenen Signals zwischen Tonhöhe bzw. Oberwellen des anderen Signals liegen.
  4. Hörgerät (10) nach Anspruch 1, wobei die Frequenzmodifizierungseinheit (52) umfasst
    eine Filterbank; und wobei die Frequenzmodifizierungseinheit (52) zum Filtern der Schätzung des Zielsignals (26) gestaltet ist, wobei Frequenzbänder der Schätzung des Zielsignals (26) zugeordnet sind, und
    Filtern der Schätzung des Masker-Signals (30), wobei andere Frequenzbänder der Schätzung des Masker-Signals (30) zugeordnet sind.
  5. Hörgerät (10) nach einem der vorangehenden Ansprüche, des Weiteren umfassend eine Phasenverschiebungsschaltung (54), die zur Phasenverschiebung einer der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30) in Bezug auf die andere der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30) gestaltet ist.
  6. Hörgerät (10) nach einem der vorangehenden Ansprüche, wobei die Signaltrennungseinheit (12) zum Bereitstellen der Schätzung auf der Basis von Spektraleigenschaften der Audiosignals gestaltet ist.
  7. Hörgerät (10) nach einem der vorangehenden Ansprüche, wobei die Signaltrennungseinheit (12) zum Bereitstellen der Schätzung auf der Basis von statistischen Eigenschaften der Audiosignals gestaltet ist.
  8. Hörgerät (10) nach einem der vorangehenden Ansprüche, wobei die Signaltrennungseinheit (12) einen Strahlformer umfasst.
  9. Hörgerät (10) nach Anspruch 8, wobei der Strahlformer adaptive Filter umfasst, die zum Filtern entsprechender Mikrofonaudiosignale und zum Anpassen der jeweiligen Filterkoeffizienten zur Minimierung der Summe der Ausgangssignale der Filter gestaltet sind.
  10. Binaurales Hörgerätsystem (10), umfassend ein erstes Hörgerät (10A) nach einem der vorangehenden Ansprüche und
    ein zweites Hörgerät (10B), umfassend mindestens ein Mikrofon (14B, 16B) zum Bereitstellen mindestens eines Mikrofonaudiosignals (18B, 20B) als Reaktion auf Ton, der bei dem mindestens einen Mikrofon (14B, 16B) empfangen wird, und wobei
    ein Sender/Empfänger (36B) in dem zweiten Hörgerät (10B) zum Übertragen von Signalen, die das mindestens eine Mikrofonaudiosignal (18B, 20B) darstellen, zu dem ersten Hörgerät (10A) angeschlossen ist und wobei
    ein Sender/Empfänger (36) im ersten Hörgerät (10A) für einen Empfang der Signale, die das mindestens eine Mikrofonaudiosignal (14B, 16B) des zweiten Hörgeräts (10B) darstellen, angeschlossen ist und wobei
    die Signaltrennungseinheit (12) zum Bereitstellen der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30) auf der Basis der mindestens einen Mikrofonaudiosignale (18, 20, 18B, 20B) des ersten und zweiten Hörgeräts (10A, 10B) gestaltet ist.
  11. Binaurales Hörgerätsystem (10) nach Anspruch 10, umfassend ein erstes Hörgerät (10A) nach Anspruch 8 oder 9, wobei der Strahlformer zum Bereitstellen der Schätzung auf der Basis von Mikrofonaudiosignalen (18, 20, 18B, 20B) des ersten und zweiten Hörgeräts (10A, 10B) gestaltet ist.
  12. Verfahren zur Signalverstärkung in einem Hörgerät (10), wobei das Verfahren die Schritte umfasst
    Bereitstellen mindestens eines Mikrofonaudiosignals (18, 20) als Reaktion auf Ton und
    Bereitstellen einer Schätzung eines Zielsignals (26) und einer Schätzung eines Masker-Signals (30) auf der Basis des mindestens einen Audiosignals (18, 20)
    Frequenzmodifizieren mindestens einer der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30), Senden einer Kombination der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30) nach Modifizierung zu einem der Trommelfelle eines Benutzers des Hörgeräts (10),
    dadurch gekennzeichnet, dass
    die Frequenzmodifizierung derart ausgeführt wird, dass nach einer Modifizierung die Schätzung des Zielsignals (26) und die Schätzung des Masker-Signals (30) im Wesentlichen in verschiedenen Frequenzbändern liegen.
  13. Verfahren zur Signalverstärkung nach Anspruch 13, umfassend die Schritte:
    Bereitstellen mindestens eines Mikrofonaudiosignals (18, 20, 18B, 20B) an beiden Ohren des Benutzers als Reaktion auf Ton, der bei beiden Ohren empfangen wird, und
    Bereitstellen der Schätzung eines des Zielsignals (26) und des Masker-Signals (30) auf der Basis der Mikrofonaudiosignale (18, 20, 18B, 20B) an beiden Ohren.
  14. Verfahren zur Signalverstärkung nach einem der Ansprüche 12 oder 13, umfassend den Schritt eines Strahlformens auf der Basis der Mikrofonaudiosignale.
  15. Verfahren zur Signalverstärkung nach Anspruch 13, umfassend den Schritt des adaptiven Filterns der Mikrofonaudiosignale durch Anpassen der jeweiligen Filterkoeffizienten, um die Summe der adaptiv gefilterten Ausgangssignale zu minimieren.
  16. Verfahren zur binauralen Signalverstärkung nach einem der Ansprüche 12 bis 15, des Weiteren umfassend den Schritt
    einer Phasenverschiebung einer der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30) in Bezug zu der anderen der Schätzung des Zielsignals (26) und der Schätzung des Masker-Signals (30).
EP20120175247 2011-12-30 2012-07-06 Hörgerät mit Frequenzdemaskierung Active EP2683179B1 (de)

Priority Applications (5)

Application Number Priority Date Filing Date Title
DK12175247.1T DK2683179T3 (en) 2012-07-06 2012-07-06 Hearing aid with frequency masking
EP20120175247 EP2683179B1 (de) 2012-07-06 2012-07-06 Hörgerät mit Frequenzdemaskierung
US13/547,720 US9185499B2 (en) 2012-07-06 2012-07-12 Binaural hearing aid with frequency unmasking
JP2012279621A JP2013153427A (ja) 2011-12-30 2012-12-21 周波数アンマスキング機能を有する両耳用補聴器
CN201210593890.2A CN103546849B (zh) 2011-12-30 2012-12-31 具有频率无掩蔽的双耳助听器

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JP4939722B2 (ja) 2000-07-14 2012-05-30 ジーエヌ リザウンド エー／エス 同期式ステレオ聴覚システム
DE102008046966B3 (de) * 2008-09-12 2010-05-06 Siemens Medical Instruments Pte. Ltd. Hörgerät und Betrieb eines Hörgeräts mit Frequenztransposition
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