EP1017253B1 - Séparation aveugle de sources pour prothèses auditives - Google Patents

Séparation aveugle de sources pour prothèses auditives Download PDF

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Publication number
EP1017253B1
EP1017253B1 EP99310611A EP99310611A EP1017253B1 EP 1017253 B1 EP1017253 B1 EP 1017253B1 EP 99310611 A EP99310611 A EP 99310611A EP 99310611 A EP99310611 A EP 99310611A EP 1017253 B1 EP1017253 B1 EP 1017253B1
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EP
European Patent Office
Prior art keywords
signal
input signals
unmixing
signals
microphones
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.)
Revoked
Application number
EP99310611A
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German (de)
English (en)
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EP1017253A2 (fr
EP1017253A3 (fr
Inventor
Justinian Rosca
Christian Darken
Thomas Petsche
Inga Holube
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Sivantos GmbH
Siemens Corp
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Siemens Corp
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Publication of EP1017253A3 publication Critical patent/EP1017253A3/fr
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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
    • H04R25/405—Arrangements for obtaining a desired directivity characteristic by combining a plurality of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/40—Arrangements for obtaining a desired directivity characteristic
    • H04R25/407—Circuits for combining signals of a plurality of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2225/00—Details of deaf aids covered by H04R25/00, not provided for in any of its subgroups
    • H04R2225/43—Signal processing in hearing aids to enhance the speech intelligibility
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00—Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/50—Customised settings for obtaining desired overall acoustical characteristics
    • H04R25/505—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
    • H04R25/507—Customised settings for obtaining desired overall acoustical characteristics using digital signal processing implemented by neural network or fuzzy logic

Definitions

  • the present invention generally relates to electronic filtering for enhancing a desired signal component of a mixed signal, and more specifically to a method and apparatus for real-time unmixing (separation or deconvolving) of a desired signal from a mixture of independent signals, particularly useful, for example, in a hearing aid.
  • noise When one is listening to someone or something, "noise" or undesired signals that interfere with the voice or desired signal, are ubiquitous. People with hearing impairment are especially vulnerable to noise. Background conversations, interference from digital devices (mobile telephones), car, or other specific environment noises, can make it very difficult for a hearing impaired person to understand a desired speech signal.
  • a reduction in the noise level of a signal, coupled with an automatic focus on a desired signal component, can significantly improve the performance of an electronic voice processor, such as one used in an advanced hearing aid.
  • WO 97/11533 discloses a directional acoustic signal processor and EP 0 883 325 discloses a method and processor for processing sounds, suitable for use in association with a hearing aid, which maximizes the signal to noise ratio of a signal from a source in an on-beam direction.
  • hearing aids using digital signal processing have been introduced. They contain one or more microphones, analog to digital converters, digital signal processors, and speakers. Usually the digital signal processors divide the incoming signals into several frequency regions using filter banks. Within each of those regions, signal gain and dynamic compression parameters can be individually adjusted in accordance with the requirement for a particular user of the hearing aid, in an attempt to improve intelligibility. Additionally, digital signal processing algorithms for feedback reduction and noise reduction are available, however they have major limitations. For example, some of the disadvantages of the currently available algorithms for noise reduction are the limited improvement they obtain when speech and background noise are in the same frequency region, due to their inability to distinguish between speech and background noise.
  • BSS Blind Source Separation
  • An electronic filtering device for performing real-time unmixing of a signal desired to be recovered by a user of the device, where the desired signal emanates from one of a plurality of independent signal sources.
  • Two microphones positioned along a common axis develop first and second electrical input signals in response to reception by the microphones of acoustic signals from the plurality of independent signal sources.
  • the spatial position of the common axis of the microphones is controllable in real time by the user to align the common axis so it points in the direction of the source of the desired signal, thereby imparting an inherent directionality to the input signals.
  • An adaptive unmixing signal processor responsive to the input signals develops output signals wherein the desired signal is separated from the mixture signal.
  • a preprocessor is provided to enhance the inherent directionality of the input signals by establishing a relative time delay therebetween. Furthermore, the preprocessor may subject the enhanced input signals to a decorrelation processing before their application to the unmixing signal processor.
  • a selected output of the unmixing signal processor can be applied as an input to a speaker for reproduction, or can be further processed for signal enhancement by an additional processor before reproduction.
  • FIG. 1 illustrates in block diagram form an application of the invention for use in hearing aids.
  • a hearing aid 10 includes two microphones 12 and 14 for developing two input signals 1 and 2, respectively.
  • the microphones are mounted in the hearing aid such that a common axis of their positioning always extends substantially in the direction in which the wearer of the hearing aid looks when being attentive to a signal source such as a voice.
  • This microphone positioning imparts an inherent directionality to input signals 1 and 2. Since each microphone develops electrical signals representative of the acoustic waves received thereby from sound sources within it's operating range, each input signal may comprise a mixture of unknown signals from an unknown number of signal sources.
  • Input signals 1 and 2 are processed in three main stages.
  • the input signals are preprocessed for enhancing the inherent directionality already imparted thereto by their positioning.
  • the resulting signals are subjected to an unmixing processing (sometimes referred to as separation processing), which is designed to produce estimates of the original unknown signals picked-up by microphones 12 and 14.
  • the outputs of the unmixing processing are preferably postprocessed to produce the desired signal 22, which can then be applied to a speaker 24 of the hearing aid 10 for reproduction and presentation to a user.
  • preprocessing stage 16 begins with normalization of the raw input signals.
  • Automatic Gain Control is used to normalize input signals 1 and 2 to a [-1,+1] range.
  • preprocessing stage 16 in order to adapt a blind source separation (BSS) technique for use in a device as small as a hearing aid, and to have it operate in real-time, preprocessing stage 16 also provides at least the first, and preferably both of the following additional processing:
  • the window D comprised 16,000 samples.
  • the above described preprocessing facilitates the subsequent BSS processing to arrive at a solution in a shorter time than if the preprocessing was not provided, and furthermore, increases the probability that the BSS processing will arrive at a valid solution instead of a local minimum.
  • Figure 3 illustrates the principles of the operation of a BSS algorithm upon which the unmixing or separation of the desired component from the input signals is based.
  • the technique is called Blind Source Separation because it makes few assumptions about the type of signals present in the mixture.
  • BSS processing is intended to recover the set of n unknown source signals from a set of their mixtures, assuming that the n source signals are independent. More specifically, as shown in Figure 3 , if s is a vector of n sources, and x is a vector of m observations of those sources (i.e., the raw input signals from the m microphones), the goal of a BSS processor is to discover the m by n mixing matrix A:
  • the sources s (s 1 , s 2 ) and the environment-dependent mixing matrix A are unknown.
  • FIG. 4 illustrates a block diagram of the main components of a BSS processor 400.
  • BSS processor 400 comprises: an unmixing component 402 for recording and updating the state of the unmixing process defined by parameters W and v; a nonlinear component 404 for generating statistics used in the adaptation process; and an adaptation component 406 for computing changes in the values of the unmixing parameters, ⁇ W and ⁇ v.
  • the BSS processor 400 continuously adapts two state variables: the 2 by 2 unmixing matrix W, and the 2 by 1 bias vector b .
  • the nonlinear component 404 transforms the output of the system using an invertible mapping.
  • the objective of component 404 is to avoid processing very large numeric values of the outputs, which may be infinities from a computational point of view. This objective is carried out by processing statistically equivalent quantities, obtained after running the outputs z through the invertible mapping.
  • the adaptation component 406 determines changes in the unmixing parameters W and v: i.e., ⁇ W and ⁇ v.
  • the objective is to maximize the mutual information that the outputs y contain about the inputs x , as well known to those skilled in this technology, and as described, for example by A.J. Bell and T.J. Sejnowski in their article entitled "An information-maximization approach to blind separation and blind deconvolution" published in Neural Computation, 7:1129-1159, 1995, and as also described in Bell's US patent 5,706,402 .
  • a typical value for the learning rate ⁇ is 0.005.
  • unmixer 18 is the postprocessing step 20, wherein a determination is made of which output estimate of unmixer 18 is more likely to represent voice rather than noise, as well as a normalization of the power of the outputs by scaling them to the level of the input powers.
  • the output signal section can be based on multiple criteria using, for example, voice specific feature extraction and analysis, and/or dominant speaker detection, which can also be accomplished using feature extraction and analysis.
  • the BSS processing is applied for use in hearing aids.
  • the inputs to the system are given by two microphones which, with the present invention, can be situated very close to one another.
  • the present invention addresses the following problems:
  • teaching of the present invention can be extremely useful for interference cancellation, separation of one voice from a mixture of many voices ("cocktail party" problem), and for preprocessing sound mixtures for noise reduction in order to allow further processing of a desired sound signal. x . All such changes, modifications, variations and other uses and applications which do not depart from the teachings herein are deemed to be covered by this patent, which is limited only by the claims which follow as interpreted in light of the foregoing description.

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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 (14)

  1. Prothèse auditive (10) incluant un dispositif de filtrage électronique pour effectuer le démixage en temps réel d'un signal qu'un utilisateur du dispositif souhaite récupérer, où le signal souhaité émane de l'une d'une pluralité de sources de signaux indépendantes, la prothèse auditive comprenant :
    un logement commun avec deux microphones (12, 14) montés à l'intérieur, le logement commun étant destiné à être partagé avec l'oreille de l'utilisateur en cours d'utilisation, et un processeur de signaux par démixage adaptatif (18), dans laquelle :
    les deux microphones (12, 14) sont positionnés le long d'un axe commun pour développer des premier et second signaux d'entrée électriques en réponse à la réception par les microphones de signaux acoustiques de la pluralité de sources de signaux indépendantes, la position spatiale de l'axe commun des microphones étant contrôlable en temps réel selon la direction dans laquelle l'utilisateur regarde lorsqu'il est attentif à une source de signaux, pour aligner l'axe commun afin qu'il pointe de façon sensiblement continue dans la direction de la source du signal souhaité lorsque l'utilisateur regarde dans la direction de la source ; et dans laquelle
    le processeur de signaux par démixage adaptatif (18) comprend un séparateur aveugle de signaux de source réceptif auxdits signaux d'entrée pour développer des signaux de sortie dans lesquels le signal souhaité est séparé du signal mélangé.
  2. Prothèse auditive selon la revendication 1, comprenant en outre un préprocesseur (16) pour modifier les signaux d'entrée avant qu'ils soient appliqués au processeur de signaux par démixage.
  3. Prothèse auditive selon la revendication 2, dans laquelle le préprocesseur (16) introduit un retard relatif entre les composantes des signaux d'entrée.
  4. Prothèse auditive selon la revendication 2 ou 3, dans laquelle le préprocesseur (16) soumet les signaux d'entrée à un traitement de décorrélation.
  5. Prothèse auditive selon la revendication 1, comprenant en outre un postprocesseur (20) réceptif aux signaux de sortie du processeur de signaux par démixage pour sélectionner le signal souhaité pour application à un dispositif de reproduction de signal.
  6. Prothèse auditive selon la revendication 1, dans laquelle le séparateur aveugle de signaux de source (18) comprend un réseau neuronal pour effectuer un processus d'apprentissage non supervisé qui sert à maximiser l'entropie en sortie jointe des signaux de sortie.
  7. Procédé pour effectuer un démixage en temps réel d'un signal qu'un utilisateur souhaite récupérer, où le signal souhaité émane de l'une d'une pluralité de sources de signaux indépendantes, le procédé comprenant les étapes suivantes :
    le positionnement de deux microphones (12, 14) le long d'un axe commun, pour développer des premier et second signaux d'entrée électriques en réponse à la réception par les microphones de signaux acoustiques de la pluralité de sources de signaux indépendantes, ledit positionnement étant tel que l'axe commun des microphones est contrôlable en temps réel par l'utilisateur pour aligner l'axe commun afin qu'il pointe de façon sensiblement continue dans la direction de la source du signal souhaité en plaçant l'axe commun à proximité de l'utilisateur d'une manière telle qu'il pointe dans la direction dans laquelle l'utilisateur regarde ; et
    la soumission desdits signaux d'entrée à un traitement de signal par démixage adaptatif en utilisant un processus de séparation aveugle de signaux sources pour développer des signaux de sortie dans lesquels le signal souhaité est séparé du signal en mélange.
  8. Procédé selon la revendication 7, dans lequel ledit positionnement place l'axe commun sur un logement commun qui est censé être partagé avec l'oreille de l'utilisateur.
  9. Procédé selon la revendication 7, comprenant en outre une étape de prétraitement pour modifier les signaux d'entrée avant qu'ils ne soient soumis au traitement de signaux par démixage.
  10. Procédé selon la revendication 9, dans lequel l'étape de prétraitement introduit un retard relatif entre les signaux d'entrée.
  11. Procédé selon la revendication 9 ou 10, dans lequel l'étape de prétraitement soumet les signaux d'entrée relativement retardés à un traitement de décorrélation.
  12. Procédé selon la revendication 11, dans lequel l'étape de traitement de décorrélation est effectuée par une diagonalisation d'une matrice de corrélation formée en utilisant les signaux d'entrée relativement retardés.
  13. Procédé selon la revendication 7, comprenant en outre une étape de post-traitement réceptive aux signaux de sortie de l'étape de traitement de signaux par démixage pour sélectionner le signal souhaité pour application à un dispositif de reproduction de signal.
  14. Procédé selon la revendication 7, dans lequel le processus de séparation aveugle de signaux sources comprend un processus d'apprentissage non supervisé qui sert à maximiser l'entropie en sortie jointe des signaux de sortie.
EP99310611A 1998-12-30 1999-12-24 Séparation aveugle de sources pour prothèses auditives Revoked EP1017253B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US22348598A 1998-12-30 1998-12-30
US223485 1998-12-30

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EP1017253A2 EP1017253A2 (fr) 2000-07-05
EP1017253A3 EP1017253A3 (fr) 2003-03-26
EP1017253B1 true EP1017253B1 (fr) 2012-10-31

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Cited By (3)

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EP3849215A1 (fr) 2020-01-10 2021-07-14 Sonova AG Transmission de deux flux audio sans fil permettant la diversité spatiale ou la capture de sa propre voix (ovpu)
US11083031B1 (en) 2020-01-10 2021-08-03 Sonova Ag Bluetooth audio exchange with transmission diversity
US11134349B1 (en) 2020-03-09 2021-09-28 International Business Machines Corporation Hearing assistance device with smart audio focus control

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US6711528B2 (en) * 2002-04-22 2004-03-23 Harris Corporation Blind source separation utilizing a spatial fourth order cumulant matrix pencil
DK1579728T3 (da) 2002-12-20 2008-02-11 Oticon As Mikrofonsystem med retningsbestemt fölsomhed
DK1509065T3 (da) 2003-08-21 2006-08-07 Bernafon Ag Fremgangsmåde til behandling af audiosignaler
JP4145323B2 (ja) 2003-09-19 2008-09-03 ヴェーデクス・アクティーセルスカプ 補聴器の受音特性の指向性制御方法および制御可能な指向特性を備える補聴器用の信号処理装置
DE10351509B4 (de) 2003-11-05 2015-01-08 Siemens Audiologische Technik Gmbh Hörgerät und Verfahren zur Adaption eines Hörgeräts unter Berücksichtigung der Kopfposition
DE102005032274B4 (de) † 2005-07-11 2007-05-10 Siemens Audiologische Technik Gmbh Hörvorrichtung und entsprechendes Verfahren zur Eigenstimmendetektion
WO2007028250A2 (fr) 2005-09-09 2007-03-15 Mcmaster University Procede et dispositif d'amelioration d'un signal binaural
EP1912472A1 (fr) * 2006-10-10 2008-04-16 Siemens Audiologische Technik GmbH Procédé pour le fonctionnement d'une prothèse auditive and prothèse auditive
DE102006047983A1 (de) 2006-10-10 2008-04-24 Siemens Audiologische Technik Gmbh Verarbeitung eines Eingangssignals in einem Hörgerät
AU2007306366B2 (en) 2006-10-10 2011-03-10 Sivantos Gmbh Method for operating a hearing aid, and hearing aid
DE102006047982A1 (de) * 2006-10-10 2008-04-24 Siemens Audiologische Technik Gmbh Verfahren zum Betreiben einer Hörfilfe, sowie Hörhilfe
JP5295115B2 (ja) * 2006-10-10 2013-09-18 シーメンス アウディオローギッシェ テヒニク ゲゼルシャフト ミット ベシュレンクテル ハフツング 補聴器の駆動方法および補聴器
DE102006047986B4 (de) * 2006-10-10 2012-06-14 Siemens Audiologische Technik Gmbh Verarbeitung eines Eingangssignals in einem Hörgerät
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CN102428716B (zh) * 2009-06-17 2014-07-30 松下电器产业株式会社 助听器装置
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CN103517941A (zh) 2011-05-06 2014-01-15 莱茵化学莱瑙有限公司 新颖的基于多羟基烷酸酯（pha）的持久水解稳定的生物基塑料、其生产方法及其用途
DE102016225207A1 (de) * 2016-12-15 2018-06-21 Sivantos Pte. Ltd. Verfahren zum Betrieb eines Hörgerätes
CN108597531B (zh) * 2018-03-28 2021-05-28 南京大学 一种通过多声源活动检测来改进双通道盲信号分离的方法

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Cited By (4)

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Publication number Priority date Publication date Assignee Title
EP3849215A1 (fr) 2020-01-10 2021-07-14 Sonova AG Transmission de deux flux audio sans fil permettant la diversité spatiale ou la capture de sa propre voix (ovpu)
US11083031B1 (en) 2020-01-10 2021-08-03 Sonova Ag Bluetooth audio exchange with transmission diversity
US11134350B2 (en) 2020-01-10 2021-09-28 Sonova Ag Dual wireless audio streams transmission allowing for spatial diversity or own voice pickup (OVPU)
US11134349B1 (en) 2020-03-09 2021-09-28 International Business Machines Corporation Hearing assistance device with smart audio focus control

Also Published As

Publication number Publication date
DK1017253T3 (da) 2013-02-11
EP1017253A2 (fr) 2000-07-05
CN1261759A (zh) 2000-08-02
EP1017253A3 (fr) 2003-03-26

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