EP2982136B1 - Procédé d'estimation d'un signal utile et dispositif auditif - Google Patents
Procédé d'estimation d'un signal utile et dispositif auditif Download PDFInfo
- Publication number
- EP2982136B1 EP2982136B1 EP14710644.7A EP14710644A EP2982136B1 EP 2982136 B1 EP2982136 B1 EP 2982136B1 EP 14710644 A EP14710644 A EP 14710644A EP 2982136 B1 EP2982136 B1 EP 2982136B1
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- EP
- European Patent Office
- Prior art keywords
- microphone
- signal
- signal vector
- vector
- reference signal
- Prior art date
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- 238000000034 method Methods 0.000 title claims description 25
- 239000013598 vector Substances 0.000 claims description 59
- 239000011159 matrix material Substances 0.000 claims description 21
- 230000000903 blocking effect Effects 0.000 claims description 9
- 230000005236 sound signal Effects 0.000 claims description 9
- 238000001914 filtration Methods 0.000 claims description 6
- 238000000926 separation method Methods 0.000 claims description 6
- 238000001228 spectrum Methods 0.000 claims description 2
- 230000009467 reduction Effects 0.000 description 5
- 230000002123 temporal effect Effects 0.000 description 4
- 230000003595 spectral effect Effects 0.000 description 3
- 210000000988 bone and bone Anatomy 0.000 description 2
- 238000001514 detection method Methods 0.000 description 2
- 238000010586 diagram Methods 0.000 description 2
- 210000000613 ear canal Anatomy 0.000 description 2
- 230000000694 effects Effects 0.000 description 2
- 238000000605 extraction Methods 0.000 description 2
- 230000006870 function Effects 0.000 description 2
- 230000004807 localization Effects 0.000 description 2
- 230000001629 suppression Effects 0.000 description 2
- 230000001052 transient effect Effects 0.000 description 2
- 208000032041 Hearing impaired Diseases 0.000 description 1
- 230000003044 adaptive effect Effects 0.000 description 1
- 230000003321 amplification Effects 0.000 description 1
- 238000004364 calculation method Methods 0.000 description 1
- 230000008859 change Effects 0.000 description 1
- 230000001419 dependent effect Effects 0.000 description 1
- 210000000883 ear external Anatomy 0.000 description 1
- 238000002474 experimental method Methods 0.000 description 1
- 230000006698 induction Effects 0.000 description 1
- 230000002452 interceptive effect Effects 0.000 description 1
- 238000003199 nucleic acid amplification method Methods 0.000 description 1
- 230000008569 process Effects 0.000 description 1
- 230000000638 stimulation Effects 0.000 description 1
- 210000003454 tympanic membrane Anatomy 0.000 description 1
Images
Classifications
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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
- H04R25/43—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
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
- H04R25/40—Arrangements for obtaining a desired directivity characteristic
- H04R25/407—Circuits for combining signals of a plurality of transducers
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L21/0216—Noise filtering characterised by the method used for estimating noise
- G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
- G10L2021/02165—Two microphones, one receiving mainly the noise signal and the other one mainly the speech signal
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L21/0216—Noise filtering characterised by the method used for estimating noise
- G10L2021/02168—Noise filtering characterised by the method used for estimating noise the estimation exclusively taking place during speech pauses
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2460/00—Details of hearing devices, i.e. of ear- or headphones covered by H04R1/10 or H04R5/033 but not provided for in any of their subgroups, or of hearing aids covered by H04R25/00 but not provided for in any of its subgroups
- H04R2460/01—Hearing devices using active noise cancellation
Definitions
- the present invention relates to a method for estimating a useful signal of a hearing device by obtaining at least two microphone signals from a sound signal, obtaining a residual signal from the microphone signals, in which a portion of the microphone signals is blocked from a predeterminable direction, and filtering the microphone signals with a filter, whereby an estimate for the useful signal is obtained.
- the present invention relates to a hearing device with a corresponding microphone device, blocking device and a filter.
- Under a hearing device is here any portable in or on the ear, a sound stimulus producing device understood, in particular a hearing aid, 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 used 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 particular challenge when using a hearing aid or another hearing device is their use in a so-called cafeteria scenario.
- the wearer of the hearing device or the hearing device converses with a conversation partner.
- the acoustic environment is still characterized by other people speaking as well as by undefined background noise.
- it is particularly difficult to extract the conversation partner's speech from the entire sound signal i. to determine or estimate the actual useful signal.
- the interference signal or noise consists of background noise and / or interference speech components or interferences.
- second order statistic quantities in particular power spectral density PSD
- the perturbation components In order for reliable estimates to be performed only during the target speech pauses, the perturbation components must be sufficiently stationary in time so that the obtained estimate is valid even if the target speaker is after a certain amount Pause is active again. In reality, however, the interfering signals are not always stationary. Therefore, effective multichannel noise reduction techniques are limited in their application because they are hardly feasible in non-stationary signal (eg speech like interference) scenarios.
- VAD Voice Activity Detection
- a method and a sound signal processing system for suppressing interference and noise in binaural microphone configurations is known.
- a coherence quantity for estimating a noise PSD matrix is determined based on reference signals in which a portion of the microphone signals received by the microphone configuration is blocked from a predeterminable direction.
- the object of the present invention is therefore to provide a method for estimating a useful signal of a hearing device, which can also be used in temporally non-stationary signals, such as speech.
- a corresponding hearing device is to be provided.
- the reference signal vector can also be one-dimensional, ie consist of a single reference signal. As a rule, however, it will consist of several reference signals.
- the reference signal vector i. From portions of the residual signal, a coherence quantity, and in particular a coherence matrix, is obtained, from which a power density quantity, and in particular a power density matrix, of the residual signal (i.e., the noise components) can be determined.
- the filter is parameterized so that a specific useful signal source can be filtered out or estimated from the microphone signals or the microphone signal vector.
- the proposed concept can also be used to estimate the spectral power densities of noise components for non-stationary signals (eg speech) so that multi-channel noise reduction techniques can be applied or realized in virtually any scenario.
- the predeterminable direction of the useful signal from the microphone signal vector is estimated in order to obtain the reference signal vector. This makes it possible to hide the useful signal from the entire space of incidence of the sound.
- each useful signal component of each microphone signal can be matched with one another and subsequently subtracted from one another.
- the signal channels one channel each for a microphone or a microphone signal
- the wanted signal components are matched to each other both in terms of delay and in terms of their spectra.
- the wanted signal components can be almost completely removed from the signal channels.
- the power density based control can be provided to the filter on the basis of the coherence quantity and the reference signal vector.
- the useful signal may in particular be a voice signal.
- the method according to the invention or the hearing device according to the invention can be used in particular to increase speech intelligibility.
- the reference signal vector may comprise speech signal components which are not part of the useful signal.
- the reference signal vector includes, for example, speech portions of speakers different from the target speaker.
- FIG. 2 The block diagram shown on the one hand represents a method which is used in a hearing aid according to FIG. 1 or can be implemented in another hearing device.
- the in FIG. 2 represented blocks corresponding devices of a hearing device.
- An exemplary hearing device or an exemplary hearing device comprises a sensor or microphone arrangement with at least two sensors or two microphones M1, Mp. In the following, representatives of microphones are always used.
- Each microphone M1, Mp converts the respectively applied sound signal into a corresponding microphone signal.
- the sound signals are components of a sound field, which represents the acoustic situation of a hearing aid wearer, for example. Such a typical situation would be that of a "cafeteria scenario" in which the hearing aid wearer is talking to a conversation partner, one or more other people are talking in the background and other background noise is present. But there may also be another acoustic situation in which non-stationary noise is given.
- the microphone signals which together form a microphone signal vector x, are each further processed in separate channels, ie a microphone signal is processed in each channel.
- this multichannel processing is represented by thick arrows.
- the microphone signal vector x is supplied to a source localization unit LOC (source localization) in the multi-channel system 10.
- LOC source localization
- the position information .phi.q of the useful signal source Sq is determined in three-dimensional space or simply as angle or distance and distance.
- This position information ⁇ q is used as coarse reference information for creating a blocking matrix BM.
- blocking matrix BM those components which originate from the spatial region of the useful signal source are spatially hidden from the microphone signals or the microphone signal vector x.
- a blocking matrix BM can for example be based on a directional blind source separation algorithm as described in US Pat Y. Zheng, K. Reindl, and W. Kellermann "BSS for improved interference estimation for blind speech signal extraction with two microphones," in IEEE International Workshop on Computational Advances in Multi-Sensor Adaptive Processing (CAMSAP) Aruba, Dutch Antilles, Dec 2009 described is. However, it is also possible to use any other algorithms for determining the blocking matrix BM.
- a multichannel reference signal or a reference signal vector n results from the microphone signal vector x. If the signals are subtracted, for example, in pairs in the blocking matrix, the number of signals of the multidimensional reference signal vector n can be half the number of microphone signals or signals. correspond to channels. With an odd number of microphone signals is preferably rounded up.
- the reference signal vector is thus usually a multidimensional vector of several individual signals.
- the reference signal vector n is supplied to a coherence estimation unit COH together with the microphone signal vector x, which consists of the individual microphone signals. It estimates a coherence matrix r from the two vectors n and x.
- the coherence matrix r is supplied to a PSD estimation unit PSD.
- the PSD estimator estimates from the coherence matrix ⁇ and the reference vector n a multi-dimensional power density estimate S, as described, for example, in FIG I. McCowan and H. Bourlard, "Microphone array postfilter for diffuse noise field," in IEEE Int. conf. Acoustics, Speech, Signal Processing (ICASSP), 2002, pp. 905-908 or in K. Reindl., Y. Zheng, A. Schwarz, S. Meier, R. Maas, A. Sehr, and W. Kellermann, "A stereophonic acoustic signal extraction scheme for noise and reverberant environments," Computer Speech and Language, 2012 ,
- a multi-channel filter FILT estimates from the power density estimate S filter parameters. These are applied in the filter FILT to the microphone signals or to the microphone signal vector x, resulting in the estimated signal s q for the particular user source or the useful signal.
- an estimation of non-stationary second order statistic quantity concerning interference components can be achieved by means of PSD, by making use of the coherence of the corresponding interference components. It can in particular the target speech components are first equated in all channels (delay compensation and spectral equalization) so that nearly identical target speech components are included in the available channels. For this approximation, a directional blind-source separation algorithm of the above type can be used. From the resulting signals, as detailed above, the spurious signal coherence matrix can be estimated, which in turn is used to estimate the spurious PSD matrix S. To estimate the useful signal according to the invention therefore requires no limitations of the temporal signal characteristics.
- the present invention utilizes that the respective acoustic scenario is spatially stationary to estimate the noise PSD matrix. It can be assumed that the spatial domain is sufficiently stationary for any scenarios, in contrast to the temporal domain. This is because the changes in the coherence function mainly depend on the spatial properties, ie the geometric arrangement of the sources and objects in the acoustic scene. The changes in the coherence function, on the other hand, are only slightly dependent on the temporal properties of the signals.
- the method according to the invention or the hearing device according to the invention is not limited to specific scenarios relating to temporally stationary noise. Accordingly, the inventive concept makes powerful, multi-channel noise reduction techniques for any scenario in which noise suppression is necessary, can be used or realized.
- An essential component of the invention is therefore based on the knowledge to separate the estimation of the spatial coherence of interference signals from the estimation of the temporal statistics quantities of the second order (PSD of the noise components).
- the space-time coherence matrices can also be continuously estimated for scenarios with (temporally) transient speech signals.
- a multi-channel Wiener filter can be used as the filter.
- a single-channel filter can also be used.
- Such filtering can be used for example in noise suppression in a binaural hearing aid.
- the PSD noise estimate along with the multi-channel Wiener filter can be implemented in conjunction with a polyphase filter bank typically used in hearing aids.
- the concept according to the invention can be realized on the basis of a signal-to-interference ratio (SIR / SINR) amplification.
- SIR / SINR signal-to-interference ratio
- an ideal blind source separation scheme is assumed for the calculation, i. the target language components are roughly the same in all available channels.
- ideal block-based voice activity detection (VAD) can be used to estimate the noise coherence matrix.
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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)
Claims (10)
- Procédé d'évaluation d'un signal utile d'un appareil auditif par- acquisition d'au moins deux signaux de microphone respectivement à partir d'un signal sonore, les signaux de microphone formant un vecteur de signal de microphone (x),- acquisition d'un vecteur de signal de référence à partir du vecteur de signal de microphone (x), vecteur de signal de référence (n) avec lequel une part des signaux de microphone provenant d'une direction pouvant être prédéfinie est bloquée, et- filtrage du vecteur de signal de microphone (x) avec un filtre (FILT), un signal d'évaluation (s̃ q) pour un signal utile étant ainsi obtenu,
caractérisé par- détermination d'une grandeur de cohérence (Γ) à partir du vecteur de signal de référence (n) et du vecteur de signal de microphone (x),- détermination d'une grandeur de densité de puissance (S) à partir de la grandeur de cohérence (Γ) et- paramétrage du filtre (FILT) à l'aide de la grandeur de densité de puissance (S). - Procédé selon la revendication 1, la direction pouvant être prédéfinie du signal utile étant estimée à partir du vecteur de signal de microphone (x) en vue d'acquérir le vecteur de signal de référence (n).
- Procédé selon la revendication 2, l'acquisition du vecteur de signal de référence (n) étant effectuée par un algorithme de séparation aveugle de source directionnel.
- Procédé selon l'une des revendications précédentes, l'acquisition du vecteur de signal de référence (n) étant effectuée en alignant l'une sur l'autre une composante de signal utile de chaque signal de microphone puis en les soustrayant ensuite l'une de l'autre.
- Procédé selon la revendication 4, les composantes de signal utile étant alignées l'une sur l'autre aussi bien du point de vue du retard que du point de vue de leurs spectres.
- Procédé selon l'une des revendications précédentes, la grandeur de cohérence (Γ) étant une matrice de cohérence.
- Procédé selon l'une des revendications précédentes, le vecteur de signal de référence (n) étant également utilisé pour la détermination de la grandeur de densité de puissance (S).
- Procédé selon l'une des revendications précédentes, le signal utile étant un signal vocal.
- Procédé selon l'une des revendications précédentes, le vecteur de signal de référence (n) comprenant des parts de signal vocal qui ne font pas partie du signal utile.
- Appareil auditif comprenant- un dispositif à microphone (M1, ..., Mp) destiné à acquérir au moins deux signaux de microphone respectivement à partir d'un signal sonore, les signaux de microphone formant un vecteur de signal de microphone (x),- un dispositif de blocage destiné à acquérir un vecteur de signal de référence (n) à partir du vecteur de signal de microphone (x), avec lequel une part des signaux de microphone provenant d'une direction pouvant être prédéfinie est bloquée, et- un filtre (FILT) destiné à filtrer le vecteur de signal de microphone (x), un signal d'évaluation (s̃ q) pour un signal utile étant ainsi obtenu,
caractérisé par- un dispositif de calcul destiné à déterminer une grandeur de cohérence (Γ) à partir du vecteur de signal de référence (n) et du vecteur de signal de microphone (x) et à déterminer une grandeur de densité de puissance (S) à partir de la grandeur de cohérence (Γ) ainsi qu'à paramétrer le filtre (FILT) à l'aide de la grandeur de densité de puissance (S).
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE102013205790.3A DE102013205790B4 (de) | 2013-04-02 | 2013-04-02 | Verfahren zum Schätzen eines Nutzsignals und Hörvorrichtung |
PCT/IB2014/059290 WO2014162214A1 (fr) | 2013-04-02 | 2014-02-27 | Procédé d'estimation d'un signal utile et dispositif auditif |
Publications (2)
Publication Number | Publication Date |
---|---|
EP2982136A1 EP2982136A1 (fr) | 2016-02-10 |
EP2982136B1 true EP2982136B1 (fr) | 2018-06-13 |
Family
ID=50288202
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP14710644.7A Revoked EP2982136B1 (fr) | 2013-04-02 | 2014-02-27 | Procédé d'estimation d'un signal utile et dispositif auditif |
Country Status (5)
Country | Link |
---|---|
US (1) | US9736599B2 (fr) |
EP (1) | EP2982136B1 (fr) |
DE (1) | DE102013205790B4 (fr) |
DK (1) | DK2982136T3 (fr) |
WO (1) | WO2014162214A1 (fr) |
Families Citing this family (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
DE102013205790B4 (de) | 2013-04-02 | 2017-07-06 | Sivantos Pte. Ltd. | Verfahren zum Schätzen eines Nutzsignals und Hörvorrichtung |
CN113380266B (zh) * | 2021-05-28 | 2022-06-28 | 中国电子科技集团公司第三研究所 | 一种微型双麦克风语音增强方法及微型双麦克风 |
Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP2395506B1 (fr) | 2010-06-09 | 2012-08-22 | Siemens Medical Instruments Pte. Ltd. | Procédé et système de traitement de signal acoustique pour la suppression des interférences et du bruit dans des configurations de microphone binaural |
EP2701145A1 (fr) | 2012-08-24 | 2014-02-26 | Retune DSP ApS | Estimation de bruit pour une utilisation avec réduction de bruit et d'annulation d'écho dans une communication personnelle |
EP2982136A1 (fr) | 2013-04-02 | 2016-02-10 | Sivantos Pte. Ltd. | Procédé d'estimation d'un signal utile et dispositif auditif |
Family Cites Families (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
KR100856246B1 (ko) * | 2007-02-07 | 2008-09-03 | 삼성전자주식회사 | 실제 잡음 환경의 특성을 반영한 빔포밍 장치 및 방법 |
EP2081189B1 (fr) * | 2008-01-17 | 2010-09-22 | Harman Becker Automotive Systems GmbH | Poste-filtre pour supports de formation de faisceau |
EP2196988B1 (fr) * | 2008-12-12 | 2012-09-05 | Nuance Communications, Inc. | Détermination de la cohérence de signaux audio |
-
2013
- 2013-04-02 DE DE102013205790.3A patent/DE102013205790B4/de active Active
-
2014
- 2014-02-27 DK DK14710644.7T patent/DK2982136T3/da active
- 2014-02-27 WO PCT/IB2014/059290 patent/WO2014162214A1/fr active Application Filing
- 2014-02-27 EP EP14710644.7A patent/EP2982136B1/fr not_active Revoked
-
2015
- 2015-10-02 US US14/873,396 patent/US9736599B2/en active Active
Patent Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
EP2395506B1 (fr) | 2010-06-09 | 2012-08-22 | Siemens Medical Instruments Pte. Ltd. | Procédé et système de traitement de signal acoustique pour la suppression des interférences et du bruit dans des configurations de microphone binaural |
EP2701145A1 (fr) | 2012-08-24 | 2014-02-26 | Retune DSP ApS | Estimation de bruit pour une utilisation avec réduction de bruit et d'annulation d'écho dans une communication personnelle |
EP2982136A1 (fr) | 2013-04-02 | 2016-02-10 | Sivantos Pte. Ltd. | Procédé d'estimation d'un signal utile et dispositif auditif |
Non-Patent Citations (2)
Title |
---|
MICHAEL SYSKIND PEDERSEN: "Source Separation for Hearing Aid Applications", KONGENS LYNBY, 2006, XP055574232, ISSN: 0909-3192 |
ULRIK KJEMS ET AL.: "MAXIMUM LIKELIHOOD BASED NOISE COVARIANCE MATRIX ESTIMATION FOR MULTI- MICROPHONE SPEECH ENHANCEMENT", 20TH EUROPEAN SIGNAL PROCESSING CONFERENCE (EUSIPCO 2012, 27 August 2012 (2012-08-27), Bucharest, Romania, pages 295 - 299, XP032254727 |
Also Published As
Publication number | Publication date |
---|---|
WO2014162214A1 (fr) | 2014-10-09 |
US9736599B2 (en) | 2017-08-15 |
DK2982136T3 (da) | 2018-09-24 |
US20160029130A1 (en) | 2016-01-28 |
DE102013205790A1 (de) | 2014-10-02 |
EP2982136A1 (fr) | 2016-02-10 |
DE102013205790B4 (de) | 2017-07-06 |
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