EP2222096A2 - Procédé destiné au traitement du signal d'un dispositif auditif et dispositif auditif correspondant - Google Patents

Procédé destiné au traitement du signal d'un dispositif auditif et dispositif auditif correspondant Download PDF

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Publication number
EP2222096A2
EP2222096A2 EP09180520A EP09180520A EP2222096A2 EP 2222096 A2 EP2222096 A2 EP 2222096A2 EP 09180520 A EP09180520 A EP 09180520A EP 09180520 A EP09180520 A EP 09180520A EP 2222096 A2 EP2222096 A2 EP 2222096A2
Authority
EP
European Patent Office
Prior art keywords
processing
signal
input signal
algorithm
processing algorithm
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Withdrawn
Application number
EP09180520A
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German (de)
English (en)
Other versions
EP2222096A3 (fr
Inventor
Matthias Latzel
Andreas Tiefenau
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Sivantos Pte Ltd
Original Assignee
Siemens Medical Instruments Pte Ltd
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Siemens Medical Instruments Pte Ltd filed Critical Siemens Medical Instruments Pte Ltd
Publication of EP2222096A2 publication Critical patent/EP2222096A2/fr
Publication of EP2222096A3 publication Critical patent/EP2222096A3/fr
Withdrawn legal-status Critical Current

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Electric hearing aids
    • H04R25/43—Electronic input selection or mixing based on input signal analysis, e.g. mixing or selection between microphone and telecoil or between microphones with different directivity characteristics
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Electric hearing aids
    • H04R25/35—Electric hearing aids using translation techniques
    • H04R25/356—Amplitude, e.g. amplitude shift or compression
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/41—Detection or adaptation of hearing aid parameters or programs to listening situation, e.g. pub, forest
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility

Definitions

  • the present invention relates to a method for processing an input signal to an output signal in a hearing device by processing the input signal according to a first processing algorithm into a first intermediate signal and processing the input signal according to a second processing algorithm into a second intermediate signal in parallel with the processing of the input signal according to the first processing algorithm , Moreover, the present invention relates to a corresponding hearing device with a first processing device and a second processing device.
  • hearing device is understood here to be any sound-emitting device that can be worn on the head or in or on the ear, in particular a hearing device, a headset, headphones and the like.
  • Hearing aids are portable hearing aids that are used to care for the hearing impaired.
  • different types of hearing aids such as behind-the-ear hearing aids (BTE), hearing aid with external receiver (RIC: receiver in the canal) and in-the-ear hearing aids (IDO), e.g. Concha hearing aids or canal hearing aids (ITE, CIC).
  • BTE behind-the-ear hearing aids
  • RIC hearing aid with external receiver
  • IDO in-the-ear hearing aids
  • ITE canal hearing aids
  • the hearing aids listed by way of example are worn on the outer ear or in the ear canal.
  • bone conduction hearing aids, implantable or vibrotactile hearing aids are also available on the market. The stimulation of the damaged hearing takes place either mechanically or electrically.
  • Hearing aids have in principle as essential components an input transducer, an amplifier and an output transducer.
  • the input transducer is usually a sound receiver, z. As a microphone, and / or an electromagnetic receiver, for. B. an induction coil.
  • the output transducer is usually as an electroacoustic transducer, z. As miniature speaker, or as an electromechanical transducer, z. B. bone conduction, realized.
  • the amplifier is usually integrated in a signal processing unit. This basic structure is in FIG. 1 shown using the example of a behind-the-ear hearing aid. In a hearing aid housing 1 for carrying behind the ear, one or more microphones 2 for receiving the sound from the environment are installed.
  • a signal processing unit 3 which is also integrated in the hearing aid housing 1, processes the microphone signals and amplifies them.
  • the output signal of the signal processing unit 3 is transmitted to a loudspeaker or earpiece 4, which outputs an acoustic signal.
  • the sound is optionally transmitted via a sound tube, which is fixed with an earmold in the ear canal, to the eardrum of the device carrier.
  • the power supply of the hearing device and in particular the signal processing unit 3 is effected by a likewise integrated into the hearing aid housing 1 battery. 5
  • a hearing aid wearer generally uses his hearing aid in different acoustic situations that place different demands on the signal processing in the hearing aid. So z. B. when listening to music a rather linear setting with little compression and regulation of the device to success, while for the understanding of language, especially in noisy environment, a rather non-linear setting, ie with compression, preferably with fast time constant benefits.
  • EP 1 829 028 A1 a method for adaptively adapting a sound processing parameter.
  • the input signal is processed to achieve a specific dynamic range.
  • a measured dynamic range is adapted to a desired dynamic range by the gain is set accordingly.
  • EP 1 307 072 A2 a hearing aid in which disturbing acoustic effects caused by switching operations should be avoided.
  • the signal processing in the hearing device leads to sliding from a first operating state to a second operating state. During the switching process both operating states are simultaneously present in the hearing aid.
  • the sliding transition occurs through parallel signal processing in at least two signal paths of the hearing aid, wherein a signal resulting from the first operating state and a signal resulting from the second operating state are added in alternating weighting.
  • the input signal is classified with regard to the current hearing situation. Depending on the classification result, the input signal is amplified according to a first compression algorithm or a second compression algorithm. As a result, the respective advantages of the various compression algorithms in the individual listening situations can be exploited.
  • the object of the present invention is to be able to better adapt the signal processing of a hearing device to a situation.
  • this object is achieved by a method for processing an input signal to an output signal in a hearing device by processing the input signal according to a first processing algorithm to a first intermediate signal and processing the input signal according to a second processing algorithm to a second intermediate signal in parallel with processing the input signal according to the first processing algorithm, and classifying the input signal and forming the output signal from both the first and the second intermediate signal a mixing ratio that depends on the result of the classification.
  • the invention provides a hearing device with a first processing device for processing an input signal according to a first processing algorithm to a first intermediate signal, a second processing device for processing the input signal according to a second processing algorithm to a second intermediate signal in parallel with the processing of the input signal according to the first processing algorithm, further comprising classifying means for classifying the input signal, and third processing means for forming an output from each of the first and second intermediate signals with a mixing ratio which depends on the result of the classifying.
  • a mixing ratio of the output signals of the processing algorithms can be controlled by the classification result.
  • a regulation preferably takes place with the level of the input signal and, in the case of the second processing algorithm, a regulation with the level of the second intermediate signal.
  • the level of the input signal for example, the compression rate, the gain or a time constant can be controlled.
  • the frequency-dependent gain can be controlled.
  • the first or the second processing algorithm may each be a compression algorithm.
  • the first processing algorithm is a linear compression algorithm (long time constant, eg 10 s) and the second processing algorithm is a non-linear compression algorithm (much shorter time constant eg 10 ms) at least in a given period of time , This allows the compression to be adapted very precisely to a specific situation.
  • the first processing algorithm may have a first time constant and the second processing algorithm may have a second time constant corresponding to the first time constant in the type, wherein the first time constant is different from the second time constant.
  • the first processing algorithm may be based on a wideband and the second processing algorithm on a narrowband level measurement.
  • both broadband signal processing and narrowband signal processing can simultaneously enter an output signal.
  • the first processing algorithm may be input-related and the second output-related.
  • FIGS. 2 to 9 detailed in the principle of the invention.
  • FIG. 2 shows a useful signal n in a first channel k1 and a second channel k2 Interference signal s.
  • the intensity I is plotted on the ordinate.
  • the useful signal n and the interference signal s thus have the drawn signal-to-noise ratio SNR1.
  • the amplified useful signal n v has compared to the amplified interference signal s v only the signal-to-noise ratio SNR2.
  • SNR2 is smaller than SNR1 (at least output-related).
  • FIG. 5 shows the respective levels L as a function of the time t.
  • the signal-to-noise ratio is SNR1.
  • FIG. 7 initially amplified by a very fast time constant, the useful portion to an amplified useful signal n v , but then the energetically lower noise applied with a higher gain, whereby an amplified interference signal s v is formed, the ratio SNR2 between useful and interference signal is degraded. This means: SNR1> SNR2. This negative effect is amplified by the use of fast time constants.
  • the compression characteristic, the time constants, the level meters used would be automatically selected by the system as a function of the situation, thus automatically ensuring the hearing aid wearer the best compression characteristic in the respective acoustic situation ,
  • a microphone 10 provides an input signal which is amplified by an amplifier 11.
  • the level of the input signal (situation) is used to control the amplifier 11.
  • the output signal of the amplifier 11 is forwarded to a handset 12.
  • the compression rate or a time constant can be controlled using the input signal. In this way, for example, a hearing program could be switched depending on the situation so that the compression parameters (amplification, compression) are adapted to the respective situation.
  • the time constants in the respective channel could be determined adaptively, in particular as a function of the (narrowband) level. This counteracts deterioration of SNR in the time domain, but spectral blurring remains a problem.
  • FIG. 9 When approaching the whole problem can also be in FIG. 9 be considered alternative presented.
  • the supplied from the microphone 10 input signal is also supplied to an amplifier 11, the output signal is passed to the handset 12.
  • the output signal is passed to the handset 12.
  • the system is similar to an AVC with the difference that the resulting frequency-specific gain is determined from a complex level statistic in several bands (eg 128). Not only purely physical but also psychoacoustic factors can be taken into account (see the above-mentioned EP 1 829 028 A1 ).
  • the system regulates very slowly and therefore operates linearly within the time constant (several seconds), it allows a pleasant sound and a pleasant loudness perception in a wide variety of acoustic environments.
  • the disadvantage of this system is that, especially in those cases in which the hearing-impaired person only has a very narrow residual dynamic (frequency-dependent difference between discomfort threshold UCL and hearing threshold HS) (eg ⁇ 30 dB), the processed signal is not complete can be mapped to the dynamic range. This has the consequence that the understanding of speech, especially in acoustic situations with background noise can only be improved insufficiently.
  • the input signal ie the signal e generated by the microphone 10
  • the input signal is in a first branch of a first Processing means 11, which is controlled with the input signal supplied.
  • a corresponding output signal a 11 is made available.
  • the input signal e from the microphone 10 is supplied to a second processing device 13, which has an output level control here. There, an output signal a 13 is generated.
  • the input signal e of the microphone 10 is finally fed to a classifier 14 in a third branch.
  • the classification result 14 is used in a weighting unit 15 to generate weights g 11 and g 13 corresponding to the output signals a 11 and a 13 .
  • the two output signals a 11 and a 13 are combined with the respective weights g 11 and g 13 so that a mixed output signal a results at the output of the weighting unit 15, which is supplied to the handset 12.
  • the compression rate, the gain or a time constant can be regulated.
  • the frequency-dependent amplification can be regulated. It is thus possible to achieve a continuous mixing of two parallel output signals a 11 , a 13 during operation, the mixing ratio depending on the classification result.
  • the gain may be 70% out of the value in a special situation AGCo and 30% from the value of AGCi.
  • AGCo Automatic Gain Control Output Dependent
  • the SNR that is important for speech understanding can be optimized in situations in which speech understanding plays a role.
  • the system can switch to a more linear system, which simultaneously sets the basic gain so that the output of the hearing aid is perceived by the individual hearing aid wearer as pleasant. If the hearing aid wearer is in a situation in which the useful signal and noise lie in different channels, the system can automatically switch completely or partially to the evaluation of the broadband power meter so that there is no different gain in the different channels and thus keep the SNR constant can be.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Neurosurgery (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Amplifiers (AREA)
EP09180520.0A 2009-01-09 2009-12-23 Procédé pour le traitement du signal dans une prothèse acoustique et prothèse acoustique correspondante Withdrawn EP2222096A3 (fr)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
DE102009004185A DE102009004185B3 (de) 2009-01-09 2009-01-09 Verfahren zur Signalverarbeitung einer Hörvorrichtung und entsprechende Hörvorrichtung

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EP2222096A2 true EP2222096A2 (fr) 2010-08-25
EP2222096A3 EP2222096A3 (fr) 2013-07-10

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EP09180520.0A Withdrawn EP2222096A3 (fr) 2009-01-09 2009-12-23 Procédé pour le traitement du signal dans une prothèse acoustique et prothèse acoustique correspondante

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US (1) US8280084B2 (fr)
EP (1) EP2222096A3 (fr)
DE (1) DE102009004185B3 (fr)

Families Citing this family (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2008116264A1 (fr) * 2007-03-26 2008-10-02 Cochlear Limited Réduction de bruit dans les prothèses auditives
US20130209970A1 (en) * 2010-02-24 2013-08-15 Siemens Medical Instruments Pte. Ltd. Method for Training Speech Recognition, and Training Device
DE102010041740A1 (de) * 2010-09-30 2012-04-05 Siemens Medical Instruments Pte. Ltd. Verfahren zur Signalverarbeitung in einem Hörhilfegerät sowie Hörhilfegerät
DK2752031T3 (en) * 2011-09-01 2015-07-27 Widex As HEARING WITH ADAPTIVE NOISE REDUCTION AND PROCEDURE

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP1307072A2 (fr) 2001-10-17 2003-05-02 Siemens Audiologische Technik GmbH Procédé pour actionner une prothèse auditive et prothèse auditive
US20070053535A1 (en) 2005-08-23 2007-03-08 Phonak Ag Method for operating a hearing device and a hearing device
DE102005061000A1 (de) 2005-12-20 2007-06-21 Siemens Audiologische Technik Gmbh Signalverarbeitung für Hörgeräte mit mehreren Kompressionsalgorithmen
EP1829028A1 (fr) 2004-12-04 2007-09-05 Dynamic Hearing Pty Ltd Procede et appareil utilisant des parametres de traitement sonore adaptatifs

Family Cites Families (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE19703228B4 (de) * 1997-01-29 2006-08-03 Siemens Audiologische Technik Gmbh Verfahren zur Verstärkung von Eingangssignalen eines Hörgerätes sowie Schaltung zur Durchführung des Verfahrens
US6236731B1 (en) * 1997-04-16 2001-05-22 Dspfactory Ltd. Filterbank structure and method for filtering and separating an information signal into different bands, particularly for audio signal in hearing aids
EP2866474A3 (fr) * 2002-04-25 2015-05-13 GN Resound A/S Méthode d'adaptation et prothèse auditive sur la base de données de perte de rapport signal/bruit

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP1307072A2 (fr) 2001-10-17 2003-05-02 Siemens Audiologische Technik GmbH Procédé pour actionner une prothèse auditive et prothèse auditive
EP1829028A1 (fr) 2004-12-04 2007-09-05 Dynamic Hearing Pty Ltd Procede et appareil utilisant des parametres de traitement sonore adaptatifs
US20070053535A1 (en) 2005-08-23 2007-03-08 Phonak Ag Method for operating a hearing device and a hearing device
DE102005061000A1 (de) 2005-12-20 2007-06-21 Siemens Audiologische Technik Gmbh Signalverarbeitung für Hörgeräte mit mehreren Kompressionsalgorithmen

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DE102009004185B3 (de) 2010-04-15
EP2222096A3 (fr) 2013-07-10
US8280084B2 (en) 2012-10-02
US20100177915A1 (en) 2010-07-15

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