EP2077059B1 - Method for operating a hearing aid, and hearing aid - Google Patents

Description

The invention relates to a method for operating a hearing aid consisting of a single or two hearing aids. Furthermore, the invention relates to a corresponding hearing aid or a corresponding hearing aid.

When one listens to someone or something, noise or unwanted acoustic signals that interfere with the voice of a counterpart or a wanted acoustic signal are omnipresent. People with a hearing loss are particularly susceptible to such noise. Conversations in the background, acoustic interference with digital devices (cell phones), car or other environmental noise can make it very difficult for a person with hearing loss to understand a desired speaker. Reducing the level of noise in an acoustic signal coupled with an automatic focus on a desired acoustic signal component can significantly improve the performance of an electronic speech processor as used in modern hearing aids.

Hearing aids with digital signal processing have been introduced in the recent past. They include one or more microphones, A / D converters, digital signal processors and speakers. Usually, the digital signal processors divide the incoming signals into a plurality of frequency bands. Within each band, signal amplification and processing may be individually adjusted in accordance with requirements for a particular hearing aid wearer to improve the intelligibility of a particular component. Furthermore, algorithms for feedback and noise minimization are available in digital signal processing, but have significant disadvantages. A disadvantage of the currently existing algorithms for noise minimization z. B. whose maximum achievable improvement in hearing aid acoustics when speech and background sounds are in the same frequency region and therefore unable to distinguish between spoken speech and background noise. (See also EP 1 017 253 A2 such as US2005 / 0265563A1 ).

This is one of the most common problems in acoustic signal processing, namely filtering out different superimposed acoustic signals of one or a plurality of them. This is also referred to as the so-called "cocktail party problem". Here, the most diverse sounds, such as music and conversations mix to an indefinable soundscape. Nevertheless, it is generally not difficult for a person to talk to a conversation partner in such a situation. It is therefore desirable for hearing aid users to be able to talk in just as situations as people without hearing loss.

In acoustic signal processing, there are spatial (eg directional microphone, beamforming), statistical (eg blind source separation) or mixed methods, which are u. a. by means of algorithms from several simultaneously active sound sources can separate a single or a plurality thereof. Thus, the blind source separation by means of statistical signal processing of at least two microphone signals allows to perform a separation of source signals without prior knowledge of their geometric arrangement. This method has advantages over conventional directional microphone approaches when used in hearing aids. In principle, with such a BSS method (BSS: Blind Source Separation) with n microphones, up to n sources can be separated, i. H. n Generate output signals.

Methods for blind source separation are known from the literature, wherein sound sources are analyzed by analyzing at least two microphone signals. Such a method and a corresponding device for this is known from EP 1 017 253 A2 known. Corresponding connecting points of the invention to the EP 1 017 253 A2 are mainly given at the end of this script.

In a special application for blind source separation in hearing aids this requires a communication of two hearing aids (analysis of at least two microphone signals (right / left)) and preferably a binaural evaluation of the signals of both hearing aids, which preferably takes place wirelessly. Alternative couplings of the two hearing aids are also possible with such an application. Such a binaural evaluation under a provision of stereo signals for a hearing aid wearer is in the EP 1 655 998 A2 taught. Corresponding connecting points of the invention to the EP 1 655 998 A2 are given at the end of this writing.

The control of directional microphones in the sense of blind source separation is subject to ambiguity, as soon as several competing sources of use, eg. B. speaker, present simultaneously. Although the blind source separation allows in principle the separation of the different sources, provided that they are spatially separated; the ambiguity, however, reduces the potential benefit of a directional microphone, although it is precisely in such scenarios that a directional microphone can be very useful for improving speech intelligibility.

The hearing aid or the mathematical algorithms for blind source separation are in principle faced with the problem of having to decide which of the signals generated by the blind source separation should be passed on most advantageously to the user of the algorithm, ie the hearing aid wearer. This is a problem that in principle is impossible for the hearing aid since the selection of the desired acoustic source directly depends on the momentary will of the hearing aid wearer and thus can not be present as an input variable for a selection algorithm. The choice made by this algorithm must be so based on assumptions about the probable will of the listener.

In the prior art, for example according to US 6 526 148 B1 , is based on a preference of an acoustic signal by the hearing aid wearer from a predetermined direction, for example, the 0 ° direction, ie the line of sight of the hearing aid wearer. This is realistic in that, in an acoustically difficult situation, the hearing aid wearer would look at his current interlocutor in order to obtain further cues which increase the speech intelligibility of the interlocutor (eg lip movements). However, this forces the hearing aid wearer to look at his interlocutor so that the directional microphone leads to increased speech intelligibility. This is especially troublesome when the hearing aid wearer wants to talk to just one person, ie is not involved in communication with multiple speakers, and does not always want / need to view his interlocutor.

However, the common assumption that the hearing aid wearer's desired acoustic source is in its 0 ° -direction is inaccurate in many cases; namely z. B. in the event that the hearing aid is standing next to his interlocutor, or sitting and with him, z. B. at the same table, other people lead a joint conversation. The hearing aid wearer would have to constantly turn his head to the side and back with a preset Wunschakustikquelle in 0 ° view, in order to follow his interlocutors.

In addition, for a "correct" or preferred by the hearing aid wearer acoustic source selection time after a source separation process so far no known technical process.

Assuming that in a communication situation, for. B. sitting at a table, not permanently a person in a 0 ° view of a hearing aid wearer is the preferred source of acoustics in the foreground, can formulate a more flexible acoustic signal selection method that is not limited by a geometric acoustic source distribution. It is therefore an object of the invention to provide an improved method for operating a hearing aid, as well as an improved hearing aid. In particular, it is an object of the invention, which output signal of a source separation, in particular a blind source separation, the hearing aid wearer is acoustically supplied. It is thus an object of the invention to find out which with high probability is a preferred acoustic source of the hearing aid wearer.

A selection of the desired Nutzakustikquelle is inventively made such that the desired speaker or the desired acoustic source is always the one or the one whose distance from a microphone (system) of the hearing preferably the least of all distances of the detected speakers or acoustic sources is. This also applies according to the invention to a plurality of speakers or acoustic sources, their distances from the microphone (system) being low in comparison with other speakers or acoustic sources.

According to the invention, a method for operating a hearing aid is provided, wherein for the tracking and selective amplification of an acoustic source, signal processing of the hearing aid for preferably all the electrical acoustic signals available to it determines a distance of the acoustic source to the hearing aid wearer and assigns it to the corresponding acoustic signal. The acoustic source or the acoustic sources with small or the shortest distances with respect to the hearing aid wearer are tracked by the signal processing and particularly taken into account in an acoustic output signal of the hearing aid.

Furthermore, a hearing aid is provided according to the invention, wherein a distance of an acoustic source to the hearing aid wearer can be determined by an acoustic module (signal processing) of the hearing aid and then assigned to electrical acoustic signals. The acoustic module then selects at least one electrical acoustic signal, which represents a small spatial distance of the associated acoustic source to the hearing aid wearer. This electrical acoustic signal is particularly considered in an output sound of the hearing aid.

In particular, the electrical acoustic signals are analyzed by the hearing aid for features that - individually or in combination - can provide information about the distance of the acoustic source to the microphone (system) or the hearing aid wearer. This preferably takes place after application of a blind source separation algorithm.

According to the invention, it is possible, depending on the number of existing microphones in the hearing aid to select a single or a plurality of (speech) acoustic sources of ambient sound and to emphasize the output sound of the hearing aid. It is possible to adjust a volume of the acoustic source or the acoustic sources in the output sound of the hearing aid as desired.

In a preferred embodiment of the invention, the signal processing has a demixing module, which preferably operates as a device for blind source separation for separating the acoustic sources of ambient sound. Furthermore, the signal processing has a post-processor module, which establishes a corresponding "near-source" operating mode in the hearing aid upon detection of a nearby acoustic source (near-acoustic-acoustic source). Furthermore, the signal processing may comprise a preprocessor module - the electrical output signals of which are the electrical input signals of the demix module - which normalizes and processes electrical acoustic signals originating from microphones of the hearing aid. Regarding the preprocessor module and the demixing module (unmixer), please refer to the EP 1 017 253 A2 Refer to paragraphs [0008] to [0023].

In a preferred embodiment of the invention, the hearing aid or the signal processing or the post-processor module leads a distance analysis of the electrical acoustic signals in that for each of the electrical acoustic signals simultaneously a distance of the corresponding acoustic source to the hearing aid is determined, and then by the signal processing or the post-processor module mainly the one or more electrical acoustic signals with a small source distance to a handset or the speaker Hearing aid are output, which converts the electrical acoustic signals into analog sound information.

Preferred acoustic sources are speech or speaker sources, whereby the selection of the speaker with the smallest horizontal distance to the ear of the hearing aid wearer - at least for many conversational situations - increases the probability that the "correct", ie. H. automatically select the voice or speaker source you want from the hearing aid user.

According to the invention, the electrical acoustic signals to be processed in the hearing aid, in particular the electrical acoustic signals separated by a source separation, are examined for information contained therein, which information can provide information about a distance of the acoustic source to the hearing aid wearer. Here, a distinction can be made between a horizontal distance and a vertical distance, wherein a too large vertical distance represents a non-preferred source. The respective distance information obtained by a single electrical acoustic signal is processed individually or in a plurality or in their entirety in such a way that a spatial distance of the acoustic source represented thereby can be determined.

In a preferred embodiment of the invention it is advantageous if the corresponding electrical acoustic signal is examined as to whether it contains spoken speech. Of particular advantage in this case is, if it is a known speaker, ie a known hearing aid Speaker whose voice profile is stored within the hearing aid with corresponding parameters.

Additional preferred embodiments of the invention will become apparent from the remaining dependent claims.

Fig. 1

ein Blockdiagramm einer Hörhilfe gemäß dem Stand der Technik, mit einem Modul für eine Blinde Quellentrennung;

Fig. 2

ein Blockdiagramm einer erfindungsgemäßen Hörhilfe mit einer erfindungsgemäßen Signalverarbeitung, bei der Verarbeitung eines Umgebungsschalls mit zwei akustisch voneinander unabhängigen Akustikquellen; und

Fig. 3

ein Blockdiagramm einer zweiten Ausführungsform der erfindungsgemäßen Hörhilfe bei der gleichzeitigen Verarbeitung von drei voneinander akustisch unabhängigen Akustikquellen des Umgebungsschalls.

The invention will be explained in more detail below with reference to embodiments with reference to the accompanying drawings. In the drawing show:

Fig. 1

a block diagram of a hearing aid according to the prior art, with a module for a blind source separation;

Fig. 2

a block diagram of a hearing aid according to the invention with a signal processing according to the invention, in the processing of an ambient sound with two acoustically independent acoustic sources; and

Fig. 3

a block diagram of a second embodiment of the hearing aid according to the invention in the simultaneous processing of three mutually acoustically independent acoustic sources of ambient sound.

The following is within the scope of the invention ( Fig. 2 & 3 ) mainly talk of a BSS module, which corresponds to a module for a blind source separation. However, the invention is not limited to such blind source separation, but rather is intended to broadly include source separation methods for acoustic signals. Therefore, this BSS module is also referred to as demixing module.

Furthermore, in the following, a "tracking" of an electrical acoustic signal by a hearing aid of a hearing aid wearer is mentioned. This is to be understood as meaning a selection of one or a plurality of electrical speech signals made by the hearing aid or a signal processing of the hearing aid or a post-processor module of the signal processing, which of the hearing aid is electrically or electronically be selected from other sources of acoustic ambient sound and which in a relation to the other acoustic sources of ambient sound amplified way, ie in a louder perceived for the hearing aid wearer, are reproduced. When the electrical acoustic signal is being tracked, a position of the hearing aid wearer in the room, in particular a position of the hearing aid in the room, ie a viewing direction of the hearing aid wearer, is preferably not taken into account by the hearing aid.

The Fig. 1 shows the state of the art, as by the EP 1 017 253 A2 (see paragraph [0008] ff) is taught. In this case, a hearing aid 1 has two microphones 200, 210, which together can form a directional microphone system, for generating two electrical acoustic signals 202, 212. Such a microphone arrangement gives the two electrical output signals 202, 212 of the microphones 200, 210 an inherent directional characteristic. Each of the microphones 200, 210 receives an ambient sound 100 that is a composite of unknown, acoustic signals from an unknown number of acoustic sources.

In the prior art, the electrical acoustic signals 202, 212 are processed primarily in three stages. In a first stage, the electrical acoustic signals 202, 212 are preprocessed in a preprocessing module 310 to improve the directional characteristic, which begins with a normalization of the original signals (equalizing the signal strength). In a second stage, a blind source separation takes place in a BSS module 320, wherein the output signals of the preprocessor module 310 are subject to a demixing process. Subsequently, the output signals of the BSS module 320 are post-processed in a post-processor module 330 to produce a desired electrical output signal 332 which serves as input to a handset 400 and loudspeaker 400 of the hearing aid 1, respectively, and a sound generated thereby to the hearing aid wearer leave. According to specification of EP 1 017 253 A2 are steps 1 and 3, Thus, preprocessor module 310 and postprocessor module 330 are optional.

Fig. 2 Now shows a first embodiment of the invention, wherein in a signal processing 300 of the hearing aid 1 a demix module 320, hereinafter referred to as BSS module 320, is located, which is a post-processor module 330 downstream. In this case, in turn, a preprocessor module 310 can be provided which appropriately prepares or prepares the input signals for the BSS module 320. The signal processing 300 is preferably carried out in a DSP (Digital Signal Processor) or in an ASIC (Application Specific Integrated Circuit).

In the following it is assumed that in the ambient sound 100 two independent acoustic 102, 104 or signal sources 102, 104 exist. One of these acoustic sources 102 is a speech source 102 arranged in relation to the hearing aid wearer, also referred to as a near acoustic source 102. The other acoustic source 104 should also be a voice source 104 in this example, but further away from the hearing aid wearer than the voice source 102. The voice source 102 should be selected and tracked by the hearing aid 1 or signal processor 300 and a primary acoustic component of the listener 400, so that an output sound 402 of the speaker 400 mainly contains this signal (102).

The two microphones 200, 210 of the hearing aid 1 each receive a mixture of the two acoustic signals 102, 104 - illustrated by the dotted arrow (representing the preferred, acoustic signal 102) and the solid arrow (representing the non-preferred, acoustic signal 104). - and deliver them either to the preprocessor module 310 or equal to the BSS module 320 as electrical input signals. The two microphones 200, 210 can be distributed as desired. They may be in a single hearing aid 1 of the hearing aid 1 or distributed to both hearing aids 1 be. In addition, it is possible, for. B. one or both microphones 200, 210 outside the hearing aid 1, z. B. on a collar or in a pen, provide as long as communication with the hearing aid 1 is guaranteed. Ie. also that the electrical input signals of the BSS module 320 need not necessarily originate from a single hearing device 1 of the hearing aid 1. Of course, more than two microphones 200, 210 for a hearing aid 1 can be realized. Preferably, a hearing aid 1 consisting of two hearing aids 1 has a total of four or six microphones.

The preprocessor module 310 prepares the data for the BSS module 320, which in turn forms two separate output signals from its two mixed input signals, depending on the capability, each of which represents one of the two acoustic signals 102, 104. The two separate output signals of the BSS module 320 are input signals for the post-processor module 330, in which it is now decided which of the two acoustic signals 102, 104 is output to the loudspeaker 400 as an electrical output signal 332.

For this (see also Fig. 3 ), the post-processor module 330 carries out a distance analysis of the electrical acoustic signals 322, 324, wherein a spatial distance to the hearing aid 1 is determined for each of these electrical acoustic signals 322, 324. Subsequently, the post-processor module 330 selects the electrical acoustic signal 322 which has the smallest distance to the hearing aid 1 and outputs this electrical acoustic signal 322 in a manner amplified relative to the other electrical acoustic signal 324 as the output electrical acoustic signal 332 (substantially corresponds to the electrical acoustic signal 322) to the Speaker 400 off.

Fig. 3 shows the inventive method and the hearing aid 1 according to the invention in the processing of three acoustic signal sources s ₁ (t), s ₂ (t), s _n (t), which together form the ambient sound 100. This ambient sound 100 is respectively of three microphones, each of which outputs an electric microphone signal x ₁ (t), x ₂ (t), x _n (t) to the signal processor 300. Here, the signal processor 300 has no preprocessor module 310, but may preferably contain this. (This also applies analogously to the first embodiment of the invention). Of course, it is also possible to process x, n acoustic sources s simultaneously via n microphones, which is indicated by the points (...) in the Fig. 3 is clarified.

The electrical microphone signals x ₁ (t), x ₂ (t), x _n (t) are input signals to the BSS module 320, which in each case in the electrical microphone signals x ₁ (t), x ₂ (t), x _n (t) acoustic sources s ₁ (t), s ₂ (t), s _n (t) separated and as electrical output signals s' ₁ (t), s' ₂ (t), s' _n (t) to the post-processor module 330.

In the following there are two speech sources s ₁ (t), s _n (t) in the vicinity of the hearing aid wearer, so that the likelihood is high that the hearing aid wearer with these two speech sources s ₁ (t), s _n (t) in a Conversation situation is. This is in the Fig. 3 also clarified by the fact that the two speech sources s ₁ (t), s _n (t) are within a voice range SR. The voice range SR should correspond to a spherical shell around the head of the hearing aid wearer, within which usual call volumes prevail. Outside the speech range SR, the corresponding volume level of a speech source s ₂ (t) is too low to assume that this speech source s ₂ (t) is in a conversation situation with the hearing aid wearer. In this case, a front half of an equatorial layer of this sphere is preferred for a conversation situation, the equatorial layer being approximately 1.5 m high, preferably 0.8-1.2 m, particularly preferably 0.4-0.7 m, and particularly preferably of 0.2-0.4m possesses. Preferably, the equator, in whose plane approximately the microphones of the hearing aid 1 are located, runs in the middle of the boundary of the equatorial layer. This can be comparatively large or comparatively different hearing aid wearers, since they often talk to a conversation partner with a vertical offset in a particular direction. Ie. for a comparatively large hearing aid wearer, the equator lies in an upper portion of the equatorial layer, so that an attention area of the hearing aid 1 is directed downwards rather than upwards. For a comparatively small hearing aid wearer, this is the other way round. This scenario is preferably suitable for a short-range area in which there is a maximum call range of 2m to 3m. Furthermore, a cylinder whose longitudinal axis coincides with a longitudinal axis of the hearing aid wearer is suitable for defining the voice range SR. For other situations, it makes more sense to define this equatorial layer over an opening angle. An opening angle is 90 ° -120 °, preferably 60 ° -90 °, in particular 45 ° -60 ° and particularly preferably 30 ° -45 °. Such a scenario is preferably suitable for a more remote area.

In the electrical acoustic signals s' ₁ (t), s' ₂ (t), s' _n (t) generated by the BSS module 320, which correspond to the speech or acoustic sources s ₁ (t), s ₂ (t), s _n (t), distance information y ₁ (t), y ₂ (t), y _n (t) are included, which can provide information about how far away the respective speech source s ₁ (t), s ₂ ( t), s _n (t) of the hearing aid 1 and the hearing aid wearer is located. The reading of this information in the form of a distance analysis takes place in the post-processor module 330, which includes a respective distance information y ₁ (t), y _{2 for} each electrical speech signal s' ₁ (t), s' ₂ (t), s' _n (t) (t), assigns y _n (t) to the acoustic source s ₁ (t), s ₂ (t), s _n (t) and then selects the one or more electrical acoustic signals s ₁ (t), s _n (t) which due to the distance information is probable that the hearing aid wearer is in a conversation situation with these speech sources s ₁ (t), s _n (t). This is in the Fig. 3 with the speech source s ₁ (t) facing the hearing aid wearer and the speech source s _n (t) at approximately a 90 ° angle next to the hearing aid wearer Hearing aid carrier is located, both of which are within the language range SR.

The post-processor module 330 now outputs the two electrical acoustic signals s ' ₁ (t), s' _n (t) to the loudspeaker 400 in an amplified manner. Furthermore, it is conceivable that z. B. the acoustic source s ₂ (t) is a source of noise and therefore ignored by the post-processor module 330, which can be determined by a corresponding module or a corresponding device in the post-processor module 330.

There are a variety of ways to determine how far an acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t) is removed from the hearing aid 1 or the hearing aid wearer, respectively, by the electrical representatives 322, 324; s' ₁ (t), s' ₂ (t), s' _n (t) of the acoustic sources 102, 104; s ₁ (t), s ₂ (t), s _n (t) evaluates accordingly.

Information about a distance between the acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t) and the hearing aid wearer, z. B. a ratio of a direct sound component to a reverb content of the corresponding acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t) and the corresponding electrical signal 322, 324; s' ₁ (t), s' ₂ (t), s' _n (t). Ie. in an individual case, the larger this ratio, the closer the acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t) on the hearing aid wearer. For this purpose, additional states within the source separation method can be analyzed that are compatible with the decision of the near acoustic source 102; s ₁ (t), s _n (t) or other acoustic source 104; s ₂ (t) precede. This is illustrated by the dashed arrow from the BSS module 320 for distance analysis in the post-processor module 330.

Further, a level criterion may indicate how far an acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t) is away from the hearing aid 1. Ie. the louder an acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t), the greater the likelihood that it is near the microphones 200, 210 of the hearing aid 1.

In addition, conclusions about the distance of an acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t). This is due to differences in a sound incidence on the left and right ear or on a left and right hearing aid 1 of the hearing aid 1.

A "punctiformity" of the source also contains information about the distance. There are methods that allow conclusions about how "punctiform" (as opposed to "diffuse") the respective acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t). In general, the more punctiform an acoustic source is, the closer it is to the microphone (system) of the hearing aid 1.

Furthermore, conclusions about a distance of the respective acoustic source 102, 104; Determine s ₁ (t), s ₂ (t), s _n (t) to the hearing aid 1. Ie. from the shape of the time signal, z. As a slope of edges of an envelope, conclusions about the distance of the corresponding acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t) are drawn.

In addition, it is of course also possible by means of a plurality of microphones 200, 210 to remove the hearing aid wearer to an acoustic source 102, 104; s ₁ (t), s ₂ (t), s _n (t) determine what z. B. can take place by triangulation.

Of course, in the second embodiment of the invention, it is also possible to amplify reproduce only a single or three or more voice acoustic sources s ₁ (t), s _n (t).

According to the invention, the distance analysis in the post-processor module 330 can always run in the background of the hearing aid 1 and, upon the occurrence of a suitable electrical speech signal 322; s ' ₁ (t), s' _n (t) are initiated. It is also possible to call the distance analysis according to the invention by the hearing aid wearer. Ie. Establishing the operating mode "near source" of the hearing aid 1 is initiated by an input device that can be called or actuated by the hearing aid wearer. Here, the input device, an operating element on the hearing aid 1 and or a control on a remote control of the hearing aid 1, z. As a button or switch, be (not shown in the figures). Moreover, it is possible to form the input device as a voice control with an associated speaker recognition module, which is tuned to a voice of the hearing aid wearer, wherein the input device is at least partially formed in the hearing aid 1 and / or at least partially in a remote control of the hearing aid 1.

In addition, it is possible to obtain additional information by means of the hearing aid 1, which of the electrical voice signals 322; s ' ₁ (t), s' _n (t) can preferably be reproduced on the hearing aid wearer as the output sound 402, s "(t) .This can be an angle of incidence of the corresponding acoustic source 102, 104; s ₁ (t), s ₂ (t ), s _n (t) to the hearing aid 1, whereby certain angles of incidence are preferred, eg, the 0 ° to ± 10 ° viewing direction (interlocutor sitting directly opposite) and / or a ± 70 ° to ± 100 ° Side direction (right / left interlocutor) and / or a ± 20 ° to ± 45 ° direction of view (interlocutor sitting diagonally opposite) of the hearing aid wearer may be preferred Furthermore, it is possible, the electrical voice signals 322; s ' ₁ (t), s' _n (t) to determine whether one of the electrical speech signals 322; s ' ₁ (t), s' _n (t) is a predominant and / or a comparatively loud electrical speech signal 322; s ' ₁ (t), s' _n (t) is and / or contains (a known) spoken language.

According to the invention, it is not necessary to carry out the distance analysis of the electrical acoustic signals 322; 324; s' ₁ (t), s' ₂ (t), s' _n (t) within the post-processor module 330. It is also possible, for. B. speed reasons, the distance analysis by another module of the hearing aid 1 to perform and the post-processor module 330 only the Selecting the one or more electrical acoustic signals 322, 324; s' ₁ (t), s' ₂ (t), s' _n (t) with the shortest distance _information (s). In such an embodiment of the invention, by definition, this other module of the hearing aid 1 is to be included in the post-processor module 330, ie, in such an embodiment, the post-processor module 330 comprises this other module.

The present document relates inter alia to a post-processor module 20 of the EP 1 017 253 A2 (Reference numeral after the EP 1 017 253 A2 ), in which by means of a distance analysis one or more speakers / acoustic sources for an electrical output signal of the post-processor module 20 are selected and reproduced therein at least amplified. See also paragraph [0025] of EP 1 017 253 A2 , Further, in the invention, the preprocessor module and the BSS module such as the preprocessor 16 and the unmixer 18 of the EP 1 017 253 A2 be constructed. See in particular paragraphs [0008] to [0024] of EP 1 017 253 A2 ,

In addition, the invention ties in with the EP 1 655 998 A2 to provide for a hearing aid wearer stereo voice signals or to enable a binaural acoustic care with speech. Here, the invention (notation according to the EP 1 655 998 A2 ) prefers the output signals z1, z2 respectively for the right (k) and left (k) of a second filter device of EP 1 655 998 A2 (please refer Fig. 2 and 3 ) for accentuation / amplification of the corresponding acoustic source downstream. Furthermore, it is also possible, the invention in the EP 1 655 998 A2 apply to the effect that it intervenes according to the teached there blind source separation and even before the second filter device. Ie. According to the invention, a selection of a signal y1 (k), y2 (k) takes place (see Fig. 3 of the EP 1 655 998 A2 ).

Claims (24)

1. Method for operating a hearing aid (1), wherein, in order to track and select a first acoustic source (102; s₁(t), s_n(t)) of an ambient sound (100; 102; 104; s₁(t), s₂(t), ..., s_n(t)), a "local source" operating mode is established by a signal processing section (300) of the hearing aid (1), wherein
   electrical acoustic signals (202, 212; 312, 314; x₁(t), x₂(t), ..., x_n(t)) are generated by the hearing aid (1) from the recorded ambient sound (100; 102; 104; s₁(t), s₂(t), ..., s_n(t)), wherein
   the electrical acoustic signals (202, 212; 312, 314; x₁(t), x₂(t), ..., x_n(t)) are unmixed into electrical output signals (322, 324; s'₁(t), s'₂(t), ..., s'_n(t)), wherein
   the first acoustic source (102; s₁(t), s_n(t)) is determined from the electrical output signals (322, 324; s'₁(t), s'₂(t), ..., s'_n(t)) and
   is selectively taken into account by the signal processing section (300) in an output sound (402; s"(t); s"₁(t)+s"_n(t)) of the hearing aid (1) such that the first acoustic source (102; s₁(t), s_n(t)) is at least acoustically prominent and is therefore better perceived compared to another acoustic source (104; s₂(t)) for a hearing aid wearer, wherein the other acoustic source (104; s₂(t)) is located spatially further away than the first acoustic source (102; s₁(t), s_n(t)) with respect to the hearing aid wearer, wherein the signal processing section (300) has, for this purpose, an unmixer module (320) for separating the electrical acoustic signals (312, 314; x₁(t), x₂(t), ..., x_n(t)) and a post-processor module (330) which establishes the "local source" operating mode of the hearing aid (1), and wherein a distance analysis of the electrical output signals (322, 324; s'₁(t), s'₂(t), ..., s'_n(t)) is performed by the signal processing section (300), and a distance (y₁(t), y₂(t), ..., y_n(t)) from the hearing aid wearer is determined for each of the acoustic sources (102, 104; s₁(t), s₂(t), ..., s_n(t)).
2. Method according to Claim 1, wherein the first acoustic source (102; s₁(t), s_n(t)) is selected in such a manner that it is located within a speaker's speech range (SR) with respect to the hearing aid wearer within which spoken language can be understood.
3. Method according to one of the preceding claims, wherein the signal processing section (300) of the hearing aid (1) is set up in such a manner that a plurality of acoustic sources (102; s₁(t), s_n(t)) that are acoustically independent of one another are tracked separately from one another.
4. Method according to one of the preceding claims, wherein the signal processing section (300) determines that or those acoustic source(s) (102; 104; s₁(t), s₂(t), ..., s_n(t)) which has/have the shortest distances (y₁(t), y₂(t), ..., y_n(t)) from the hearing aid wearer and which is/are made available to the hearing aid wearer by the output sound (402; s"(t); s"₁(t)+s"_n(t)) of the hearing aid (1).
5. Method according to one of the preceding claims, wherein the first acoustic source (102; s₁(t), s_n(t)) or the electrical output signal (322, s'₁(t), s'_n(t)) is also identified and selected on the basis of:

   • its ratio of a direct sound to the echo component;

   • a level criterion;

   • a head shadow effect;

   • a punctiformity of the respective source;

   • a time feature, in particular a form of a time signal;

   • a distance measurement based on multi-microphone processing;

   • a freedom from interference;

   • a vertical distance from the hearing aid (1) or the hearing aid wearer; and/or

   • spoken language contained therein.
6. Method according to one of the preceding claims, wherein acoustic sources (104; s₂(t)) which do not contain any speech or acoustic sources (104; s₂(t)) which are excessively disturbed by interference signals are preferably not taken into account by the signal processing section (300).
7. Method according to one of the preceding claims, wherein the unmixer module (320) is in the form of a blind source separation module (320).
8. Method according to one of the preceding claims, wherein a volume of the electrical output signals (322, 324; s'₁(t), s'₂(t), ..., s'_n(t)) is also adjusted to one another for an electrical output acoustic signal (332) from the signal processing section (300) in the post-processor module (330).
9. Method according to one of the preceding claims, wherein the signal processing section (300) has a preprocessor module (310) which conditions the electrical acoustic signals (202, 212; x₁(t), x₂(t), ..., x_n(t)) for the unmixer module (320).
10. Method according to one of the preceding claims, wherein, when establishing the "local source" operating mode, the first acoustic source (102; s₁(t), s_n(t)) comes from a particular direction with respect to the hearing aid wearer, preferably from a 0° viewing direction or a 90° lateral direction, and is then tracked by the signal processing section (300).
11. Method according to one of the preceding claims, wherein a first acoustic source (102; s₁(t), s_n(t)) which is predominant in the ambient sound (100; 102; 104; s₁(t), s₂(t), ..., s_n(t)) is tracked as the first acoustic source (102; s₁(t), s_n(t)) in the "local source" operating mode.
12. Method according to one of the preceding claims, wherein the "local source" operating mode is set up in such a manner that only the first or substantially only the first acoustic source(s) (102; s₁(t), s_n(t)) of the ambient sound (100; 102; 104; s₁(t), s₂(t), ..., s_n(t)) is/are perceived by the hearing aid wearer in the output sound (402; s"(t); s"₁(t)+s"_n(t)) of the hearing aid (1).
13. Hearing aid for tracking and selecting a first acoustic source (102; s₁(t), s_n(t)) of an ambient sound (100; 102; 104; s₁(t), s₂(t), ..., s_n(t)), wherein
    the hearing aid (1) generates electrical acoustic signals (202, 212; 312, 314; x₁(t), x₂(t), ..., x_n(t)) from the ambient sound (100; 102; 104; s₁(t), s₂(t), ..., s_n(t)) and has a signal processing section (300) for establishing a "local source" operating mode, wherein
    the signal processing section (300) has an unmixer module (320) for separating the electrical acoustic signals (202, 212; 312, 314; x₁(t), x₂(t), ..., x_n(t)) into electrical output signals (322, 324; s'₁(t), s'₂(t), ..., s'_n(t)),
    wherein the first acoustic source (102; s₁(t), s_n(t)) is determined from the electrical output signals (322, 324; s'₁(t), s'₂(t), ..., s'_n(t)),
    wherein a distance analysis of electrical output signals (322, 324; s'₁(t), s'₂(t), ..., s'_n(t)) is performed by means of the signal processing section (33), wherein a distance (y₁(t), y₂(t), ..., y_n(t)) from the hearing aid wearer is determined for the acoustic sources (102, 104; s₁(t), s₂(t), ..., s_n(t)),
    and a post-processor module (330) of the signal processing section (300) identifies and selects the first acoustic source (102; s₁(t), s_n(t)) which indicates a shorter distance from a hearing aid wearer than another acoustic source (104; s₂(t)), wherein
    the first acoustic source (102; s₁(t), s_n(t)) can be selectively taken into account in an output sound (402; s" (t) ; s"₁(t)+s"_n(t)) of the hearing aid (1) such that it is at least prominent and is therefore acoustically better perceived by the hearing aid wearer compared to another acoustic source (104; s₂(t))
14. Hearing aid according to Claim 13, wherein the signal processing section (300) of the hearing aid (1) is set up in such a manner that a plurality of acoustic sources (102; s₁(t), s_n(t)) that are acoustically independent of one another can be tracked separately from one another.
15. Hearing aid according to one of Claims 13 to 14, wherein the signal processing section (300) can be used to determine that or those acoustic source(s) (102; 104; s₁(t), s₂(t), ..., s_n(t)) which has/have the shortest distances (y₁(t), y₂(t), ..., y_n(t)) from the hearing aid wearer and which can be made available to the hearing aid wearer by the output sound (402; s"(t); s"₁(t)+s"_n(t)) of the hearing aid (1).
16. Hearing aid according to one of Claims 13 to 15, wherein the first acoustic source (102; s₁(t), s_n(t)) or the electrical output signal (322, s'₁(t), s'_n(t)) can also be identified and selected on the basis of:

    • its ratio of a direct sound to the echo component;

    • a level criterion;

    • a head shadow effect;

    • a punctiformity of the respective source;

    • a time feature, in particular a form of a time signal;

    • a distance measurement based on multi-microphone processing;

    • a freedom from interference;

    • a vertical distance from the hearing aid (1) or the hearing aid wearer; and/or

    • spoken language contained therein.
17. Hearing aid according to one of Claims 13 to 16, wherein electrical output signals (324; s'₂(t)) which do not contain any speech or electrical output signals (324; s'₂(t)) which are greatly disturbed by interference signals preferably cannot be taken into account by means of the signal processing section (300).
18. Hearing aid according to one of Claims 13 to 17, wherein the post-processor module (330) tracks and selects the electrical output signal(s) (322; s'₁(t), s'_n(t)) and generates a corresponding electrical output signal (332) for a loudspeaker (400) of the hearing aid (1), which loudspeaker outputs the output sound (402; s"(t); s"₁(t)+s"₃(t)) of the hearing aid (1).
19. Hearing aid according to one of Claims 13 to 18, wherein two hearing devices (1) of the hearing aid (1) or an individual hearing aid (1) has/have a plurality of microphones (200, 210) which can be used to record the ambient sound (100; 102; 104; s₁(t), s₂(t), ..., s_n(t)) which contains the first acoustic source (102; s₁(t), s_n(t)), and an electrical acoustic signal (202, 212; x₁(t), x₂(t), ..., x_n(t)) can be respectively output to the signal processing section (300) by means of the microphones (200, 210).
20. Hearing aid according to one of Claims 13 to 19, wherein the "local source" operating mode can be established by the post-processor module (330) of the hearing aid (1).
21. Hearing aid according to one of Claims 13 to 20, wherein the unmixer module (320) is in the form of a blind source separation module (320).
22. Hearing aid according to one of Claims 13 to 21, wherein a volume of the electrical output signals (322, 324; s'₁(t), s'₂(t), ..., s_n(t)) can also be tuned to one another for the electrical output acoustic signal (332) from the signal processing section (300) in the post-processor module (330).
23. Hearing aid according to one of Claims 13 to 22, wherein the signal processing section (300) has a preprocessor module (310) which can be used to condition the electrical acoustic signals (202, 212; x₁(t), x₂(t), ..., x_n(t)) for the unmixer module (320).
24. Hearing aid according to one of Claims 13 to 23, wherein the hearing aid (1) comprises a single hearing device or two hearing devices (1).

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