EP4304205A1 - Method for directional signal processing for a hearing instrument - Google Patents

Abstract

The invention relates to a method for directional signal processing for a hearing instrument (1), wherein a first input signal (E1) and a second input signal (E2) are generated from a sound signal (4) of the environment by a first input transducer (M1) and a second input transducer (M2) of the hearing instrument (1), wherein a first front intermediate signal (Z1v) and a first rear intermediate signal (Z1h) are formed on the basis of the first input signal (E1) and the second input signal (E2), and wherein, in particular frequency band by frequency, a first superposition (U1) of the first front intermediate signal (Z1v) and the first rear intermediate signal (Z1h) is formed by means of a complex-valued first superposition parameter (a1), and is adapted on the basis of the first superposition parameter (a1). It is provided here that a complex value (a1.0) of the first superposition parameter (a1) resulting from said adaptation of the first superposition (U1) is converted into a first alternative parameter (ap1) and a second alternative parameter (ap2), wherein at least the second alternative parameter (ap2) has an at least semicircular monotonic relationship to an angle (θ) of minimum sensitivity of the first superposition (U1), wherein the angle (θ) of minimum sensitivity is limited to a predetermined angular range (Δθ) via a corresponding limitation of the second alternative parameter (ap2), and a limited second alternative parameter (ap2') is formed in the process, and wherein an output signal (out) is generated based on the first alternative parameter (ap1) and the limited second alternative parameter (ap2') and based on a superposition of the first input signal (E1) and the second input signal (E2).

Description

The invention relates to a method for directional signal processing for a hearing instrument, wherein a first or second input signal is generated from a sound signal from the environment by a first or second input transducer of the hearing instrument, with a first front intermediate signal and a first front intermediate signal being generated based on the first and second input signals a first rear intermediate signal is formed, wherein, in particular frequency band-wise, a first overlay of the first front intermediate signal and the first rear intermediate signal is formed by means of a first overlay parameter, and is adapted based on the first overlay parameter, and wherein based on a value of the first overlay parameter and based on a In particular, an output signal is generated by time-delayed superimposition of the first input signal and the second input signal.

In hearing instruments such as hearing aids for treating a wearer's hearing loss, a corresponding number of input signals are generated from ambient sound by a number of input transducers such as microphones, which represent the air pressure fluctuations of the ambient scarf at the respective input transducer. Based on the input signal(s), an output signal is generated by signal processing, which is converted into an output sound signal by an output transducer of the hearing instrument (e.g. a loudspeaker). The signal processing can preferably be tailored to the audiological requirements of the wearer (e.g. a hearing impairment), and in particular include frequency band-wise amplification and/or compression.

In the case of two (or more) input transducers in a hearing instrument, directional processing of the input signals generated in this way can also take place. As a result, a directional signal can be generated, for example as an intermediate signal when generating the output signal, which is aimed at an assumed useful signal source (usually a conversation partner or similar) and/or which suppresses sources of interference by spatially "fading out".

Such a masking out can take place by means of a time-delayed superimposition of the two input signals, or by means of two different such superimpositions, for example by means of a so-called cardioid and an anticardioid signal, which in turn are adaptively superimposed. A potential problem here, however, is that suppressing a source of interference as completely as possible requires a signal level that is as identical as possible in the two (or more) input transducers of the hearing instrument. As a result of shading effects both from the head (or parts of the outer ear) of the wearer and from the housing of the hearing instrument, this is often not the case, which is why an input signal for completely masking out a directional source of interference must be adjusted accordingly using an angle-dependent amplification factor. However, such an amplification factor is often difficult to determine. Furthermore, such an amplification factor can also lead to strong fluctuations in a useful signal, which is undesirable. Another challenge is the additional requirement that the said suppression of the source of interference often has to be limited to a certain angular range with respect to the wearer's field of vision (e.g. to the rear half-space).

The invention is therefore based on the object of specifying a method for directional signal processing for a hearing instrument, which is as robust as possible against different signal levels of the individual input signals involved, and which allows an efficient restriction of an angle of the minimum sensitivity of a resulting directional signal.

The stated object is achieved according to the invention by a method for directional signal processing for a hearing instrument, wherein by a first Input transducer of the hearing instrument, a first input signal is generated from a sound signal from the environment, a second input signal being generated from the sound signal from the environment by a second input converter of the hearing instrument, and using the first input signal and the second input signal in each case, and preferably by a particularly time-delayed one Superposition, a first front intermediate signal and a first rear intermediate signal are formed.

It is provided here that, in particular frequency band-wise, a first superposition of the first front intermediate signal and the first rear intermediate signal is formed by means of a complex-valued first superposition parameter, and is adapted based on the first superposition parameter, with a complex value of the first resulting from said adaptation of the first superposition Overlay parameter is converted into a corresponding pair of real-valued alternative parameters, consisting of a first alternative parameter and a second alternative parameter, wherein at least the second alternative parameter has an at least semicircle-wise monotonic relationship to an angle of minimum sensitivity of the first overlay, the angle minimum sensitivity is modified via a corresponding modification of the second alternative parameter, and thereby a modified second alternative parameter is formed, and an output signal is generated based on the first alternative parameter and the modified second alternative parameter as well as based on a superposition of the first input signal and the second input signal becomes. Advantageous and partly inventive embodiments are the subject of the subclaims and the following description.

A hearing instrument generally includes any device that is set up to generate at least two corresponding input signals by means of at least two input transducers, and to use these to generate an output signal through appropriate processing, which is converted into an output sound signal by an output transducer and presented to the hearing of one Carrier is supplied to this device. In particular, this can be done as: Hearing instrument may include headphones equipped with the appropriate input transducers (e.g. as an “earplug”), a headset, data glasses with loudspeakers, etc. However, a hearing instrument also includes a hearing aid in the narrower sense, i.e. a device for treating a hearing impairment of the wearer, in which the input signals generated from ambient sound by means of the input transducers are processed into said output signal depending on the audiological requirements of the wearer and in particular amplified depending on the frequency band and/or compressed, so that the output sound signal is suitable for at least partially compensating for the wearer's hearing impairment, in particular in a user-specific manner.

A (particularly electroacoustic) input transducer includes any device which is intended and set up to generate a corresponding electrical signal from the sound signal of the environment (the associated input signal), the voltage or current fluctuations of which preferably influence the fluctuations in the air pressure of the sound signal represent and reproduce within the scope of the respective resolution. In particular, a microphone is included as an input converter.

In particular, the angle of minimum sensitivity is limited as a modification to a predetermined angular range via a corresponding limitation of the second alternative parameter, and a limited second alternative parameter is formed as a modified second alternative parameter, using the first alternative parameter and the limited second alternative parameter as well an output signal is generated based on a superposition of the first input signal and the second input signal.

Based on a modification, in particular a limitation, of the first alternative parameter, a minimum sensitivity (i.e. in particular a depth of a so-called “notch”) at the corresponding angle can advantageously be modified.

The formation or generation of a resulting signal based on one or more incoming signals is to be understood in particular as meaning that the respective signal components of the incoming signals, in particular frequency band-wise, enter the relevant resulting signal in accordance with a mapping rule, so that preferably a monotonic, particularly preferably linear There is a relationship between the amplitudes and/or envelopes and/or signal levels of the incoming signals and the corresponding size of the resulting signal.

The first front intermediate signal and the first rear intermediate signal are preferably each generated from the first and second input signals using mapping rules that are symmetrical to one another, in particular as time-delayed superpositions, so that the directional characteristics of the two first intermediate signals, based on the free space, are symmetrical to one another. However, the first front intermediate signal and the first rear intermediate signal can also be generated using mapping rules that are different from one another (in particular non-symmetrical), with the two intermediate signals mentioned preferably being linearly independent of one another. In particular, it is conceivable that one of the intermediate signals has an omnidirectional directional characteristic.

wobei Z1v und Z1h jeweils das erste vordere bzw. hintere Zwischensignal bezeichnet, $\underset{\_}{\mathrm{a}1}\in ℂ$

den ersten Überlagerungsparameter, sowie ω und t ein Frequenz- bzw. ein diskreter Zeitindex sind.The first overlay U1 is formed in particular in the form $$\mathrm{U}1\left(\mathrm{\omega },\mathrm{t}\right)=\mathrm{Z}1\mathrm{v}\left(\mathrm{\omega },\mathrm{t}\right)+\underset{\_}{\mathrm{a}1}\left(\mathrm{\omega },\mathrm{t}\right)\cdot \mathrm{Z}1\mathrm{H}\left(\mathrm{\omega },\mathrm{t}\right),$$

where Z1v and Z1h respectively denote the first front and rear intermediate signals, $\underset{\_}{\mathrm{a}1}\in ℂ$

the first superposition parameter, and ω and t are a frequency and a discrete time index, respectively.

The adaptation of the first overlay based on the first overlay parameter includes in particular that the first overlay (the actual overlay, for example according to equation (i), is used here synonymously with the signal resulting from said overlay) with respect to a parameter such as the total energy, the total level or a deviation from a reference signal or the like is optimized via the first overlay parameter, whereby the optimization can also be carried out numerically in several steps, so that the first overlay parameter (even for a given time index) via the adaptation towards one Value converges (which can be determined, for example, based on a limit value for a step size between two adaptation steps).

The value of the first overlay parameter, which generally has a real and imaginary part, is now converted into a pair of real-valued, alternative parameters, i.e. a first alternative parameter and a second alternative parameter, the latter having a monotonic relationship to an angle of minimum sensitivity has first overlay.

This can be motivated in particular by the following consideration: For a preferably stationary sound signal, which strikes the hearing instrument with respect to its frontal direction (defined in particular based on the direction from the second to the first input transducer) at an angle of θ, the relative transfer function from the first to the second input transducer is (i.e. the amplitude and phase difference as a result of the propagation of the sound signal from the sound source at the angle θ to the second instead of the first input transducer) A _θ · e ^{-iωτ cos θ} , where A _θ is an angle-dependent amplitude factor (which, among other things, accounts for shading effects caused by the wearer's head or taken into account by the housing of the hearing instrument). With a suitable choice of the two alternative parameters, in particular through a relation between the value of the first overlay parameter and the said relative transfer function, an at least semicircle-wise monotonic relationship can be formed between the second alternative parameter and the angle of minimum sensitivity of the first overlay.

existiert, sodass die besagte monotone Beziehung wenigstens für einen Winkelbereich von [ψ, ψ + π] gilt.An at least semicircle-wise monotonic relationship in particular includes that the relationship between the second alternative parameter and the angle of minimum sensitivity applies at least to an angular range of said angle which covers at least a semicircle, ie that a $\mathrm{\psi }\in ℝ$

exists, so that the said monotonic relationship applies at least for an angular range of [ψ, ψ + π].

Using the second alternative parameter, this angle can now be reduced to a desired, predetermined angular range, i.e. to the rear half-space (θ ∈ [90°, 270°]), or a narrower "wedge" in the rear half-space (e.g. θ ∈ [120°, 240°]), by limiting the value range of the alternative second parameter to a corresponding interval (and if necessary, a sign of the said angle θ with respect to the frontal or 180° direction is taken into account).

By limiting the angle of minimum sensitivity to the predetermined angular range, the alternative second parameter is adapted to the interval of the value range of this second alternative parameter corresponding to the said angle restriction, whereby in particular the limited second alternative parameter is generated. This limited second alternative parameter can preferably be identical to the second alternative parameter if the associated angle θ of the minimum sensitivity is already within the predetermined angular range, or otherwise be given by a limit value of such an interval.

An output signal is now generated based on this limited second alternative parameter and a superposition of the first and second input signals. This can be done in particular by reversing the calculation of the two alternative parameters from the first overlay parameter in such a way that an adapted first overlay parameter is formed based on the first alternative parameter and the limited second alternative parameter, and accordingly the superposition of the two input signals is given to generate the output signal is caused by the first superposition (which, as a result of its creation, consists of the first front and first back Intermediate signal also represents a superposition of the two input signals), but now with the first adapted superposition parameter.

mit dem Vektor der Eingangssignale E ^T = (E1, E2) und dem Koeffizientenvektor w = (w1, w2)^T, wobei die Koeffizienten w1 und w2 von der konkreten Form der Erzeugung des ersten vorderen und hinteren Zwischensignals Z1v, Z1h in Gleichung (i) abhängen.The output signal can be converted directly by an output transducer of the hearing instrument (such as a loudspeaker) into an output sound signal, which is fed to the hearing of the wearer of the hearing instrument. Alternatively, the output signal of the method can go through further signal processing steps (such as further noise suppression and/or frequency band amplification or compression) before the output sound signal is generated from it. In particular, a further signal can be added to the output signal before conversion into the output scarf signal. Such a pair of alternative parameters can be formed in particular by representing the first superposition U1 according to equation (i) by a corresponding conversion in the basis of the two input signals as $$\mathrm{U}1=\mathrm{E}1\cdot \mathrm{w}1+\mathrm{E}2\cdot \mathrm{w}2={\mathbf{E}}^{\mathrm{T}}\cdot \mathbf{w}$$

with the vector of input signals E ^T = (E1, E2) and the coefficient vector w = (w1, w2) ^T , where the coefficients w1 and w2 depend on the specific form of generation of the first front and rear intermediate signals Z1v, Z1h in equation (i ) depend.

In order to gain a better understanding of the coefficient vector w , the underlying adaptation of the first superposition U1 is assigned to a 1-order finite impulse response filter (FIR filter), which then carries out a type of “spatial sampling” of the sound signal. The associated filter polynomial is $$\mathrm{P}\left(\mathrm{e.g}\right)=\mathrm{w}1+\mathrm{w}2\cdot {\mathrm{e.g}}^{-1}.$$

eindeutig bestimmt. Entsprechend weisen Signale, welche sich von der ersten Überlagerung U1 um einen solchen skalaren Vorfaktor $\underset{\_}{\mathrm{c}}\in ℂ$

unterscheiden, hinsichtlich ihrer Richtwirkung dieselben Eigenschaften auf wie diese.The zeros of the polynomial in equation (iii) are z0 = - w2/w1, and are complex except for a prefactor $\underset{\_}{\mathrm{c}}\in ℂ$

clearly determined. Correspondingly, signals that differ from the first superposition U1 have such a scalar Prefactor $\underset{\_}{\mathrm{c}}\in ℂ$

differ, have the same properties as these in terms of their directivity.

In view of the said ambiguity of the zero point of the FIR filter according to equation (iii), which is assigned to the adaptation of the first superposition, the coefficient vector is now set to w ₀ = c · [1, - r · e ^iϕ ], where the relative phase ϕ and the quotient r of the amounts of the two coefficients w2/w1, as already mentioned, depend on the specific design of the first front and rear intermediate signals Z1v, Z1h. The first alternative parameter can now be formed based on the said quotient r (which therefore indicates the ratio of the amounts of the coefficients), and in particular as this, the second alternative parameter based on the relative phase ϕ of the coefficients to one another, and in particular as this.

bzw. $$\left[1,{\mathrm{A}}_{\mathrm{\theta }}\cdot {\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }\cos \mathrm{\theta }}\right]\cdot {\left[1,-\mathrm{r}\cdot {\mathrm{e}}^{\mathrm{i\varphi }}\right]}^{\mathrm{T}}=0.$$

This can be seen if, as mentioned above, a stationary sound signal is applied from an angle θ, which must be attenuated as completely as possible. With the relative transfer function between the two input converters already mentioned above, the vector E of the two input signals results as E = E1 · h with h = [1, A _θ · e ^{-iωτ cos θ} ] ^T . The sound signal then requires cancellation in the present illustration $${\mathbf{H}}^{\mathrm{T}}\cdot {\mathbf{w}}_0=0,$$

or. $$\left[1,{\mathrm{A}}_{\mathrm{\theta }}\cdot {\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }\cos \mathrm{\theta }}\right]\cdot {\left[1,-\mathrm{r}\cdot {\mathrm{e}}^{\mathrm{i\varphi }}\right]}^{\mathrm{T}}=0.$$

A solution resulting from equation (vi`) is r = 1/A _θ , ϕ = ωt · cos θ. This makes it clear that for θ ∈ [0, π] there is a monotonic relationship between the relative phase ϕ of the two coefficients w1, w2 of the first superposition (in the representation of the input signals) and the angle of the minimum sensitivity. This angle can now be limited using the relative phase ϕ as a second alternative parameter.

It proves to be further advantageous if the value of the first overlay parameter is converted into a corresponding real-valued second one Overlay parameter and an associated value of a real-valued gain factor, wherein the real-valued gain factor is assigned to a corresponding gain of the second input signal when forming the first front or rear intermediate signal, and the second overlay parameter is adjusted such that for a second overlay, which is formed based on of the second overlay parameter from the first front intermediate signal and the first and rear intermediate signals while amplifying the second input signal with said amplification factor, the angle of minimum sensitivity is limited to the predetermined angular range, and thereby an adapted second overlay parameter is generated, and that based on the adapted second Superposition parameter and the gain factor and based on a particularly time-delayed superposition of the first input signal and the second input signal, the output signal is generated.

entspricht dabei einer Verstärkung des zweiten Eingangssignals bei der Bildung des ersten vorderen bzw. ersten hinteren Zwischensignals. Mit anderen Worten werden anhand des ersten Überlagerungsparameters $\underset{\_}{\mathrm{a}1}\in ℂ$

(für die erste Überlagerung des ersten vorderen und des ersten hinteren Zwischensignals) also der Verstärkungsfaktor m und der zweite Überlagerungsparameters $\mathrm{a}2\in ℝ$

derart ermittelt, dass die erste Überlagerung dabei übergeht in eine zweite Überlagerung des ersten vorderen und des ersten hinteren Zwischensignals, wobei bei besagten ersten Zwischensignalen jeweils das zweite Eingangssignal zuvor mit dem Verstärkungsfaktor m verstärkt, also skaliert wurde, und wobei die zweite Überlagerung dieser Zwischensignale gebildet wird anhand des zweiten Überlagerungsparameters $\mathrm{a}2\in ℝ$

. Im Allgemeinen ist diese Umrechnung von Real- und Imaginärteil von $\underset{\_}{\mathrm{a}1}\in ℂ$

nach $\left(\mathrm{a}2,\mathrm{m}\right)\in {ℝ}^2$

wohldefiniert.The said real gain factor $\mathrm{m}\in ℝ$

corresponds to an amplification of the second input signal when forming the first front or first rear intermediate signal. In other words, based on the first overlay parameter $\underset{\_}{\mathrm{a}1}\in ℂ$

(for the first superposition of the first front and the first rear intermediate signal) i.e. the gain factor m and the second superposition parameter $\mathrm{a}2\in ℝ$

determined in such a way that the first overlay merges into a second overlay of the first front and first rear intermediate signals, with the second input signal previously being amplified, i.e. scaled, with the amplification factor m in said first intermediate signals, and the second overlay of these intermediate signals being formed is based on the second overlay parameter $\mathrm{a}2\in ℝ$

. In general, this conversion of real and imaginary part of $\underset{\_}{\mathrm{a}1}\in ℂ$

after $\left(\mathrm{a}2,\mathrm{m}\right)\in {ℝ}^2$

well-defined.

It is not absolutely necessary that said second superposition (i.e. the resulting signal) is actually formed (approximately analogous to equation (i)); Rather, it is sufficient to simply convert a1 → (a2, m) according to the restrictions resulting in particular from the intermediate signals (e.g. for the amplification factor).

eine minimale Empfindlichkeit bzw. maximale Abschwächung aufweist. Hierbei bezeichnen in Gleichung (i') Z2v und Z2h ein zweites vorderes bzw. zweites hinteres Zwischensignal, welche jeweils aus dem ersten vorderen bzw. ersten hinteren Zwischensignal durch eine vorige Verstärkung des zweiten Eingangssignals mit dem besagten Verstärkungsfaktor hervorgehen.Ideally, the gain factor is now determined in such a way that a superposition of the first intermediate signals (with the corresponding previous application of the gain factor to the second input signal when forming the intermediate signals) enables a source of interference to be completely masked out, and thus the function of a level adjustment between the two input converters of the hearing instrument. In this case, a monotonic relationship can now generally be established between the second overlay parameter a2 and the angle for which the above-mentioned second overlay $$\mathrm{U}2\left(\mathrm{\omega },\mathrm{t}\right)=\mathrm{Z}2\mathrm{v}\left(\mathrm{\omega },\mathrm{t}\right)+\mathrm{a}2\left(\mathrm{\omega },\mathrm{t}\right)\cdot \mathrm{Z}2\mathrm{H}\left(\mathrm{\omega },\mathrm{t}\right),$$

has a minimum sensitivity or maximum attenuation. Here, in equation (i'), Z2v and Z2h denote a second front and second rear intermediate signal, respectively, which each arise from the first front and first rear intermediate signal by a previous amplification of the second input signal with the said amplification factor.

).As already described, a monotonic relationship can now be established between the angle of a maximum attenuation of the second overlay U2 and the second overlay parameter a2 (however, the monotonicity is here only defined over angular ranges of half a revolution, i.e. for angle θ ∈ [γ, γ + π ] with $\mathrm{\gamma }\in ℝ$

).

The second overlay parameter can form the second alternative parameter, and the gain factor can form the first alternative parameter. However, the first alternative parameter can also be formed, as described using equation (iv`), based on the quotient r of the amounts of the two coefficients w1, w2 of the first superposition with respect to the two input signals, and the second alternative parameter based on the relative phase ϕ of the two coefficients to each other, with the adjustment of the second

Overlay parameter is carried out based on said alternative parameters r and ϕ, and so the adapted second overlay parameter is formed.

An output signal is now generated based on this adapted second superposition parameter and a superposition of the first and second input signals. The said superposition of the first and second input signals can in particular be given by the second superposition (of the second front and second rear intermediate signals) according to equation (i'), whereby the adapted second superposition parameter a2' (instead of the "original" second superposition parameter a2 ) is to be used. When generating the output signal, a correction filter for the frequency response can in particular be added in order to ensure a flat frequency response in the frontal direction (defined, for example, by the direction from the second to the first input transducer of the hearing instrument). In the event that the time delay in the relevant overlays is implemented using a frequency factor, the correction filter can in particular also be given by a frequency-dependent correction factor.

In particular, based on the first input signal and the second input signal scaled by means of the real-valued amplification factor, a second front intermediate signal and a second rear intermediate signal are formed, preferably by a particularly time-delayed superimposition,
wherein the output signal is generated based on the second overlay using the adjusted second overlay parameter.

However, the said superposition of the two input signals to generate the output signal can also be given by the first superposition according to equation (i), but the amplification factor m and the adjusted second superposition factor a2 'back to the then "adjusted" value for the first superposition parameter a1 'is mapped, in particular using the reverse mapping rule (a2', m) → a1' .

Preferably, the first rear intermediate signal is formed in such a way that it has a relative attenuation in a frontal direction, which is defined in particular based on a direction from the second input transducer to the first input transducer, and the first front intermediate signal is formed in such a way that it is in a direction opposite to the frontal direction shows a relative weakening. In particular, the first front and the first rear intermediate signals are symmetrical to one another. In particular, this also applies to the second front and second rear intermediate signals. A relative attenuation is to be understood in particular as a local and preferably global minimum of sensitivity across all angles. This minimum does not necessarily have to mean a maximum attenuation in the sense of total masking out, but can in particular also assume finite values for the respective sensitivity for the first intermediate signals.

Conveniently, the first front intermediate signal and the first rear intermediate signal are each generated using a time-delayed superposition of the two input signals, with the second input signal for the first front intermediate signal and the first input signal for the first rear intermediate signal, preferably around the acoustic transit time between the two Input converters, is delayed. As a result, directional signals are generated as the first intermediate signals, which have a cardioid or anticardioid-shaped directional characteristic in free space and are particularly suitable for the present method due to the simple yet stable generation.

Expediently, at least in one frequency band, preferably up to a band limit frequency of up to 500 Hz, a delay between the input signals, in particular in the time-frequency domain, is implemented by means of a particularly additional all-pass filter. In the time-frequency domain, a delay can be implemented via a phase factor, which depends on the center frequency of the frequency band in question. However, depending on the implementation, this center frequency for the first frequency band can be 0 Hz, so that no delay would be possible. In this case, an alternative implementation of the delay via an all-pass filter is favorable. This can but can also be advantageous for other, low frequency bands if the phase within a frequency band has large changes, which are only inadequately represented with a phase factor that is constant over the frequency band in question.

It proves to be further advantageous if, in a first adaptation step, a first value of the complex first overlay parameter is determined, said first value of the first overlay parameter is converted into the corresponding first and second alternative parameters, and the limited second alternative parameter is determined from this, a second value of the first overlay parameter is determined based on the first alternative parameter and the limited second alternative parameter, and said second value of the first overlay parameter is used for a second adaptation step. In other words, it is not necessary that the limitation of the angular range for the angle of minimum sensitivity of the second overlay does not occur after the adaptation of the first overlay parameter has been completely completed. Rather, such a restriction can also occur in a single adaptation step, and the limited second alternative parameter can form the basis for the next adaptation step.

Advantageously, the first overlay parameter is determined using a least-mean-squares algorithm and/or using a gradient method. These methods mentioned are particularly suitable for adapting the complex-valued first overlay parameter with real and imaginary parts, i.e. in particular for optimizing the associated first overlay with regard to a parameter via the first overlay parameter. The gradient method can in particular include an application of a gradient of the real and imaginary part with respect to such a parameter (such as a signal level or a deviation from an error or reference signal).

The invention further mentions a hearing instrument, comprising a first input transducer for generating a first input signal from a sound signal from the environment, a second input transducer for generating a second Input signal from the sound signal of the environment, and a control unit, wherein the hearing instrument is set up to carry out the method described above. The hearing instrument is set up, in particular by means of the control unit, to carry out the method steps in which one of the input signals or signals derived therefrom is processed. For this purpose, the control unit is in particular equipped with at least one signal processor.

The hearing instrument according to the invention shares the advantages of the method according to the invention. The advantages stated for the method and its further development can be transferred analogously to the hearing instrument.

Fig. 1

in einer Draufsicht Richtcharakteristiken von Zwischensignalen eines Hörinstrumentes,

Fig. 2

in einer Draufsicht die Richtcharakteristiken der Zwischensignale nach Fig. 1 im Fall ungleicher Signalpegel der Eingangswandler,

Fig. 3

in einem Blockdiagramm den Ablauf eines Verfahrens zur direktionalen Signalverarbeitung in einem Hörinstrument, und

Fig. 4

in einem Blockdiagram eine zum Verfahren nach Fig. 3 alternative Ausgestaltung.

An exemplary embodiment of the invention is explained in more detail below with reference to drawings. Here we show schematically:

Fig. 1

in a top view, directional characteristics of intermediate signals of a hearing instrument,

Fig. 2

The directional characteristics of the intermediate signals are shown in a top view Fig. 1 in the case of unequal signal levels of the input converters,

Fig. 3

in a block diagram the sequence of a method for directional signal processing in a hearing instrument, and

Fig. 4

in a block diagram one about the procedure Fig. 3 alternative design.

Corresponding parts and sizes are provided with the same reference numbers in all figures.

In Figure 1 is shown schematically in a top view of directional characteristics for a hearing instrument 1. The hearing instrument 1 is designed as a hearing aid 2, which is intended and set up to treat a hearing impairment. The hearing instrument 1 has a first input transducer M1 and a second

Input transducer M2, which are arranged at a distance d from one another, and in the present case are each provided by corresponding microphones. A first input signal E1 is generated from a sound signal 4 from the environment by the first input transducer M1, and a second input signal E2 is generated by the second input transducer M2. Furthermore, the hearing instrument 1 has a control unit 5, which is set up to process the said input signals E1, E2, and for this purpose in particular comprises a signal processor (not shown in detail).

bzw. $$\mathrm{Z}1\mathrm{v}\left(\mathrm{\omega },\mathrm{t}\right)=\mathrm{E}1\left(\mathrm{\omega },\mathrm{t}\right)={\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\mathrm{E}2\left(\mathrm{\omega },\mathrm{t}\right)$$

Based on a time-delayed superimposition of the first input signal E1 and the second input signal E2, a first front intermediate signal Z1v is generated, the time delay corresponding exactly to the acoustic transit time of the distance d: $$\mathrm{Z}1\mathrm{v}\left(\mathrm{\omega },\mathrm{t}\right)=\mathrm{E}1\left(\mathrm{\omega },\mathrm{t}\right)-\mathrm{E}2\left(\mathrm{\omega },\mathrm{t}-\mathrm{\tau }\right),$$

or. $$\mathrm{Z}1\mathrm{v}\left(\mathrm{\omega },\mathrm{t}\right)=\mathrm{E}1\left(\mathrm{\omega },\mathrm{t}\right)={\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\mathrm{E}2\left(\mathrm{\omega },\mathrm{t}\right)$$

In the ideal case that the signal levels of the first and second input signals E1, E2 are identical (and in particular there are no shading effects and no attenuation over the distance d), the first front intermediate signal has a cardioid-shaped directional characteristic (dashed line). In a manner comparable to equation (v) or (v`), but with delay of the first input signal E1, a first rear intermediate signal Z1h = e ^-iωτ E1 - E2 is generated. In the ideal case already mentioned, the first rear intermediate signal Z1v has an anticardioid-shaped directional characteristic (dotted line), which has its maximum attenuation in a frontal direction 6. The direction of maximum attenuation of the first front intermediate signal Z1v is opposite to the frontal direction 6.

eine erste Überlagerung U1 gemäß Gleichung (i) gebildet, wobei der Wert des ersten Überlagerungsparameters a1 (also sein Real- und Imaginärteil) durch eine Adaption der ersten Überlagerung U1 bestimmt werden, etwa durch eine Minimierung der Signalenergie bzw. des Pegels mittels eines Gradientenverfahrens. Eine Störquelle 8, welche zum Schallsignal 4 der Umgebung einen gerichteten Störschall 10 beiträgt, kann nun mittels der ersten Überlagerung U1 "ausgeblendet" werden, wie die Richtcharakteristik der ersten Überlagerung U1 (durchgezogene Linie) zeigt. Diese Richtcharakteristik weist beim Winkel θ, in welchem gerade die Störquelle 8 liegt, die maximale Abschwächung auf.The first front and the first rear intermediate signals are now converted using a complex-valued first superposition parameter $\underset{\_}{\mathrm{a}1}\in ℂ$

a first overlay U1 is formed according to equation (i), where the value of the first overlay parameter a1 (i.e. its real and imaginary parts) is formed by an adaptation of the first overlay U1 can be determined, for example by minimizing the signal energy or level using a gradient method. A source of interference 8, which contributes a directed noise 10 to the sound signal 4 of the environment, can now be "hidden" by means of the first overlay U1, as the directional characteristic of the first overlay U1 (solid line) shows. This directional characteristic has the maximum attenuation at the angle θ, at which the source of interference 8 is located.

However, if the signal level for the two input signals E1, E2 is not the same due to shading effects (e.g. due to the head and/or pinna of the wearer of the hearing instrument 1, but also due to the housing of the hearing instrument 1), then depending on the type These shading effects mean that, for example, the attenuation for the first rear intermediate signal Z1h in the frontal direction 6 is no longer complete, but has a finite value. A comparable situation can apply to the first front intermediate signal Z1v depending on the specific level differences of the input signals E1, E2. As a result, the first overlay U1 in the direction of the interference source 8 may no longer be able to achieve complete attenuation and thus no longer completely mask out the interference sound 10.

This situation is shown schematically in a top view Figure 2 shown. The first front intermediate signal Z1v (dashed line) and the first rear intermediate signal Z1h now each have a directional characteristic which no longer allows complete attenuation in some directions. For this reason, the first superposition U1 (not shown), formed according to equation (i) based on the first front and rear intermediate signals Z1v, Z1h Figure 2 , in the present case not suitable for completely masking out the source of interference 8.

$$\mathrm{Z}1\mathrm{h}={\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\mathrm{E}1-\mathrm{E}2.$$

To solve this problem, a method is proposed, which is based on a block diagram in Figure 3 is pictured. In Figure 3 the sound signal 4 follows the environment Figure 1 , which includes the interference sound 10 from the directed interference source 8 (not shown in each case), through the first and second Input converter M1, M2 converted into the first and second input signals E1, E2. The first front and first rear intermediate signals Z1v, Z1h are formed from the two input signals E1, E2 by time-delayed superimposition (see description for Figure 1 , especially equation (ii`)): $$\mathrm{Z}1\mathrm{v}=\mathrm{E}1-{\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\mathrm{E}2,$$

$$\mathrm{Z}1\mathrm{H}={\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\mathrm{E}1-\mathrm{E}2.$$

In general, the signal levels of the first and second input signals E1, E2 are not the same, so that the first front and first rear intermediate signals Z1v, Z1h have directional characteristics comparable to those in Figure 2 have shown.

wird nun aus dem ersten vorderen und dem ersten hinteren Zwischensignal Z1v, Z1h eine erste Überlagerung U1 gemäß Gleichung (i) gebildet. Diese erste Überlagerung U1 wird einer Adaption 12 unterzogen, in welcher ein konkreter Wert a1.0 für den ersten Überlagerungsparameter a1 ermittelt wird. Die Adaption 12 kann dabei z.B. in einer Minimierung der Signalenergie der ersten Überlagerung U1 durch ein Gradientenverfahren bezüglich des Real- und Imaginärteils des ersten Überlagerungsparameters a1 erfolgen o.ä.Using a complex first overlay parameter $\underset{\_}{\mathrm{a}1}\in ℂ$

A first superposition U1 is now formed from the first front and the first rear intermediate signals Z1v, Z1h according to equation (i). This first overlay U1 is subjected to an adaptation 12, in which a specific value a1.0 is determined for the first overlay parameter a1. The adaptation 12 can, for example, be carried out by minimizing the signal energy of the first overlay U1 using a gradient method with respect to the real and imaginary parts of the first overlay parameter a1, or similar.

$$\mathrm{U}1={\mathbf{E}}^{\top }\cdot \mathbf{w}$$

mit dem Vektor der Eingangssignale E ^T = (E1, E2) und dem Koeffizientenvektor w = (w1, w2)^T, wobei die konkrete Form der Koeffizienten w1 und w2 nun durch Gleichung (vii) gegeben ist.According to equations (i), (v") and (vi), the first superposition U1 results in: $$\begin{array}{c}\mathrm{U}1=\mathrm{E}1\cdot \left(1+\underset{\_}{\mathrm{a}1}{\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\right)-\mathrm{E}2\cdot \left({\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}+\underset{\_}{\mathrm{a}1}\right)\\ =\mathrm{E}1\cdot \mathrm{w}1+\mathrm{E}2\cdot \mathrm{w}2,\end{array}$$

$$\mathrm{U}1={\mathbf{E}}^{\top }\cdot \mathbf{w}$$

with the vector of input signals E ^T = (E1, E2) and the coefficient vector w = (w1, w2) ^T , where the concrete form of the coefficients w1 and w2 is now given by equation (vii).

ergibt. Die relative Phase ϕ ergibt sich dabei schlicht aus dem Argument der rechten Seite von Gleichung (viii), der Faktor r ist gegeben durch den Quotienten r der Beträge der Koeffizienten w2/w1 nach Gleichung (vii). Letzterer wird nun verwendet als ein erster alternativer Parameter ap1, die relative Phase ϕ als ein zweiter alternativer Parameter ap2. Diese kann nun gemäß der sich aus Gleichung (iv`) ergebenden Beziehung e^{iϕ - iωτ cosθ} verwendet werden, um einen Winkelbereich Δθ für den Winkel θ zu begrenzen, wodurch sich eine begrenzte relative Phase ϕ' bzw. ein begrenzter alternativer zweiter Parameter ap2' ergibt. Insbesondere kann diese begrenzte relative Phase ϕ' zur relativen Phase ϕ identisch sein, wenn der Winkel θ der minimalen Empfindlichkeit der ersten Überlagerung U1 bereits im gewünschten Winkelbereich Δθ (etwa dem hinteren Halbraum bzgl. der Frontalrichtung 6) liegt.For the further procedure, the coefficient vector w according to equation (vii) is brought into the form w ₀ according to equation (iv'), where $$r{e}^{i\mathrm{\phi }}=\frac{e^{-i\mathrm{\omega \tau }}+\underset{\_}{a1}}{1+\underset{\_}{a1}e-i\mathrm{\omega \tau }}$$

results. The relative phase ϕ results simply from the argument on the right-hand side of equation (viii), the factor r is given by the quotient r of the amounts of the coefficients w2/w1 according to equation (vii). The latter is now used as a first alternative parameter ap1, the relative phase ϕ as a second alternative parameter ap2. This can now be used according to the relationship e ^{iϕ - iωτ cosθ} resulting from equation (iv`) to limit an angular range Δθ for the angle θ, which results in a limited relative phase ϕ' or a limited alternative second parameter ap2' results. In particular, this limited relative phase ϕ' can be identical to the relative phase ϕ if the angle θ of the minimum sensitivity of the first overlay U1 is already in the desired angular range Δθ (approximately the rear half-space with respect to the frontal direction 6).

The adjusted relative phase ϕ' also results in a corresponding adjustment of the first superposition parameter a1 in equation (viii). Equation (viii) can then be solved for this adjusted first superposition parameter a1 ' $$\underset{\_}{\mathrm{a}1\prime }=\frac{{\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}-\mathrm{r}{e}^{\mathrm{i}\mathrm{\phi }\prime }}{\mathrm{r}{e}^{\mathrm{i}\mathrm{\phi }\prime -\mathrm{i}\mathrm{\omega \tau }}-1}.$$

Based on the adapted first overlay U1' (dashed line) with said adapted first overlay parameter a1 ', an output signal Out can now be generated, the adapted first overlay U1' being multiplied, among other things, by a correction factor c _cor to correct the frequency response, so that In the frontal direction 6 the frequency response of the output signal Out is flat. In addition, further signal processing steps 20 such as noise or feedback suppression etc., but also frequency band-dependent enhancement depending on the wearer's audiological specifications or similar, can be interposed.

und einen reellwertigen Verstärkungsfaktor $\mathrm{m}\in ℝ$

abgebildet, wobei letztgenannter dem zweiten Eingangssignal E2 zugeordnet ist.In Figure 4 is an alternative embodiment of the method using a block diagram Figure 3 shown. As in that case, the first overlay U1 is formed based on the first overlay parameter a1, and the value a1.0 of the first overlay parameter a1 is determined in the adaptation 12. In a next step, the value a1 .0 of the first overlay parameter a1 is converted to a real-valued second overlay parameter $\mathrm{a}2\in ℝ$

and a real-valued gain factor $\mathrm{m}\in ℝ$

shown, the latter being assigned to the second input signal E2.

In order to be able to determine the relationship between the first overlay parameter a1 (or from its value a1.0 ) and the specific values of the second overlay parameter a2 and the gain factor m, a second front intermediate signal Z2v and a second rear intermediate signal Z2h are defined (dashed line Signal path), in which, however, the second input signal E2 is each subjected to the amplification factor m, i.e $$\begin{array}{l}\mathrm{Z}2\mathrm{v}=\mathrm{E}1-\mathrm{m}\cdot {\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\mathrm{E}2,\\ \mathrm{Z}2\mathrm{H}={\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\mathrm{E}1-\mathrm{m}\cdot \mathrm{E}2.\end{array}$$

This amplification factor m can be used to compensate for a different signal level between the first and second input signals E1, E2. Even in the case of different signal levels, the second front and second rear intermediate signals Z2v, Z2h therefore have the in Figure 1 shown directional characteristics, which no longer apply to the first front and first rear intermediate signals Z1v, Z1h in the general case (i.e. not in free space, but in the case of shading effects, etc.) (for this general case, these directional signals demonstrate directional characteristics as described Figure 2 on).

$$\mathrm{U}2={\mathbf{E}}^{\top }\cdot \mathbf{w}\prime .$$

One now forms from said second front and second rear intermediate signals Z2v, Z2h (which differ from the corresponding first front and first rear intermediate signals Z1v, Z1h by the said amplification factor m in the proportion of the second input signal E2) using the second Overlay parameter a2 has a second overlay U2 (dashed signal path) analogous to equation (i), the following applies to this: $$\begin{array}{c}\mathrm{U}2=\mathrm{E}1\cdot \left(1+\mathrm{a}2{\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}\right)-\mathrm{m}\cdot \mathrm{E}2\cdot \left({\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}+\mathrm{a}2\right)\\ =\mathrm{E}1\cdot \mathrm{w}1\prime +\mathrm{E}2\cdot \mathrm{w}2\prime ,\mathrm{and}\mathrm{therefore}\end{array}$$

$$\mathrm{U}2={\mathbf{E}}^{\top }\cdot \mathbf{w}\prime .$$

The amplification factor m and the second superposition parameter a2 should be determined in such a way that the representation limits the angle θ of the maximum attenuation (see Figure 1 ) should be enabled to a desired angle range.

In a manner analogous to equations (vii') and (viii), equation (x), with the consideration motivated by equations (iii) and (iv'), now becomes the connection between the relative phase ϕ and the coefficient quotient r = |w2' / w1'| on the one hand, and the gain factor m as the first alternative parameter ap1 and the second superposition parameter a2 as the second alternative parameter ap2 on the other hand: $$\mathrm{r}{e}^{i\mathrm{\phi }}=m\frac{e^{-i\mathrm{\omega \tau }}+a2}{1+a2e-i\mathrm{\omega \tau }}$$

definiert sind, wodurch w2/w1 = w2'/w1' folgt. Da der Bruch auf der rechten Seite vom Betrag 1 ist, ergibt sich für den Verstärkungsfaktor m = r, wobei r anhand des Wertes a1.0 des ersten Überlagerungsparameters a1 durch den Betrag von Gleichung (viii) gegeben ist.This took advantage of the fact that the zeros of the polynomial in equation (iii) are only up to one factor $\underset{\_}{\mathrm{c}}\in ℂ$

are defined, whereby w2/w1 = w2'/w1' follows. Since the fraction on the right-hand side is 1 in magnitude, the gain factor is m = r, where r is given by the magnitude of equation (viii) based on the value a1 .0 of the first superposition parameter a1 .

For the second superposition parameter a2 as the second alternative parameter ap2 of the method Figure 4 results from equation (xi): $$\mathrm{a}2=\frac{{\mathrm{e}}^{-\mathrm{i}\mathrm{\omega \tau }}-e^{\mathrm{i\phi }}}{{\mathrm{e}}^{\mathrm{i}\mathrm{\phi }-\mathrm{i}\mathrm{\omega \tau }}-1}=\frac{cOs\mathrm{\phi }-\mathit{cos}\left(\mathrm{\omega \tau }\right)}{1-\mathit{cos}\left(\mathrm{\phi }-\mathrm{\omega \tau }\right)}$$

Using the corresponding tabulated values, it is now possible to determine the relationship between ϕ (the relative phase of the coefficients w1' and w2' in equation (x)) and the angle θ of the minimum sensitivity of the first superposition U1 (e ^{iϕ - iωτ cosθ} = 1, see equation (iv')) and thus also the second overlay U2 (which initially only represents a conversion of the first overlay U1).

From this, a corresponding adapted value for the second overlay parameter a2, i.e. an adapted second overlay parameter a2' or a limited second alternative parameter ap2', can be determined.

The output signal out can now be formed (if necessary after further signal processing steps 20 and correction factors of the frequency response, not shown) from the second superposition U2 according to equation (i ') with the second front and second rear intermediate signals Z2v, Z2h according to equation (ix), however based on the adjusted second overlay parameter a2' (instead of, as in equation (i'), based on the second overlay parameter a2). The amplification factor m in the second front and second rear intermediate signals Z2v, Z2h according to equation (ix) is r = m according to equation (xi) with r according to equation (viii) from the first superposition parameter a1 .

However, the amplification factor m and the adapted second overlay parameter a2' can also be calculated back into the domain of the first overlay parameter a1 (not shown), so that the output signal out is then formed in that case from a first overlay based on the adjusted first overlay parameter a1 determined in this way '. This procedure has the advantage that a prefactor in the output signal out, which corrects a high-pass behavior in the frequency response of the first overlay U1, is independent of the angle θ of the minimum sensitivity.

Although the invention has been illustrated and described in detail by the preferred embodiment, the invention is not disclosed by the Limited examples and other variations may be derived therefrom by those skilled in the art without departing from the scope of the invention.

Reference symbol list

1

Hearing instrument

2

hearing aid

4

Sound signal (of the environment)

5

Control unit

6

Frontal direction

8th

Source of interference

10

noise

12

Adaptation

20

Signal processing steps

a1(')

(adjusted) first overlay parameter

a2(')

(adjusted) second overlay parameter

ap1, ap2

first or second alternative parameter

ap2`

limited second alternative parameter

ccor

Correction factor

E1, E2

first or second input signal

M1, M2

first or second input converter

out

Output signal

m

Gain factor

r

Quotient (the amounts of the coefficients)

U1, U2

first and second overlay, respectively

w1('), w2(')

Coefficients

Z1v, Z1h

first front or first rear intermediate signal

Z2v, Z2h

second front or second rear intermediate signal

Δθ

Angle range

θ

Angle (minimum sensitivity)

τ

Time Delay

ϕ

relative phase (of the coefficients)

Claims (17)

Method for directional signal processing for a hearing instrument (1), - wherein a first input signal (E1) is generated from a sound signal (4) from the environment by a first input transducer (M1) of the hearing instrument (1), - wherein a second input signal (E2) is generated from the sound signal (4) of the environment by a second input transducer (M2) of the hearing instrument (1), - whereby a first front intermediate signal (Z1v) and a first rear intermediate signal (Z1h) are formed based on the first input signal (E1) and the second input signal (E2), - wherein, in particular frequency band-wise, a first superposition (U1) of the first front intermediate signal (Z1v) and the first rear intermediate signal (Z1h) is formed by means of a complex-valued first superposition parameter ( a1 ), and is adapted based on the first superposition parameter ( a1 ), - wherein a complex value ( a1.0 ) of the first overlay parameter ( a1 ) resulting from said adaptation of the first overlay (U1) is converted into a corresponding pair of real-valued alternative parameters, consisting of a first alternative parameter (ap1) and a second alternative Parameter (ap2), wherein at least the second alternative parameter (ap2) has an at least semicircle-wise monotonic relationship to an angle (θ) of minimum sensitivity of the first overlay (U1), - wherein the angle (θ) of minimum sensitivity is modified via a corresponding modification of the second alternative parameter (ap2), thereby forming a modified second alternative parameter, and - An output signal (out) is generated based on the first alternative parameter (ap1) and the modified second alternative parameter as well as on the basis of a superposition of the first input signal (E1) and the second input signal (E2). Method according to claim 1,
wherein the angle (θ) of minimum sensitivity is limited as a modification to a predetermined angular range (Δθ) via a corresponding limitation of the second alternative parameter (ap2), and a limited second alternative parameter (ap2 ') is formed as a modified second alternative parameter, and - An output signal (out) is generated based on the first alternative parameter (ap1) and the limited second alternative parameter (ap2`) as well as on the basis of a superposition of the first input signal (E1) and the second input signal (E2). Method according to claim 1 or claim 2,
wherein a minimum sensitivity at the corresponding angle (θ) is modified based on a modification, in particular a limitation, of the first alternative parameter (ap1). Method according to one of the preceding claims, wherein a coefficient vector ( w ) of the coefficients (w1, w2) of the first input signal (E1) and the second input signal (E2) is formed in the first superposition (U1), where the first alternative parameter (ap1) is formed based on a quotient (r) of the amounts of the two coefficients (w1, w2), and where the second alternative parameter (ap2) is formed based on a relative phase (ϕ) of the two coefficients (w1, w2) to one another. Method according to claim 4, wherein an adapted first overlay parameter ( a1') is formed based on the first alternative parameter (ap1) and the limited second alternative parameter (ap2`), and wherein the overlay for generating the output signal (out) is formed by the first overlay (U1) based on the adapted first overlay parameter (a1' ). Method according to one of the preceding claims, - wherein the value of the first overlay parameter ( a1 ) is converted into a corresponding real-valued second overlay parameter (a2) and an associated value of a real-valued amplification factor (m), whereby the real-valued amplification factor (m) corresponds to a corresponding amplification of the second input signal (E2) at the Formation of the first front or rear intermediate signal (Z1v, Z1h) is assigned, - wherein the second overlay parameter (a2) is adapted in such a way that for a second overlay (U2), which is formed based on the second overlay parameter (a2) from the first front intermediate signal (Z1v) and the first rear intermediate signal (Z1h) with amplification of the second input signal (E2) with said amplification factor (m), the angle (θ) of minimum sensitivity is limited to the predetermined angular range (Δθ), and thereby an adapted second superposition parameter (a2`) is generated, and - whereby the output signal (out) is generated based on the adapted second superposition parameter (a2`) and the amplification factor (m) as well as on the basis of a superposition of the first input signal (E1) and the second input signal (E2). Method according to claim 5,
where the second superposition parameter (a2) is used as the second alternative parameter (ap2) and the gain factor (m) is used as the first alternative parameter (ap1). Method according to claim 6 in conjunction with claim 4, wherein the first alternative parameter (ap1) is formed based on the quotient (r) of the amounts of the two coefficients (w1, w2) of the first input signal (E1) and the second input signal (E2) in the first superposition (U1), where the second alternative parameter (ap2) is formed based on the relative phase (ϕ) of the two coefficients (w1, w2) to one another, and wherein the adaptation of the second overlay parameter (a2) takes place based on the first alternative parameter (ap1) and the limited second alternative parameter (ap2'), and the adapted second overlay parameter (a2`) is thus formed. Method according to one of claims 6 to 8,
wherein the output signal (out) is generated based on the first overlay (U1), the value ( a1.0 ) of the adapted first overlay parameter ( a1 ) being determined based on the adapted second overlay parameter (a2`) and on the basis of the amplification factor (m). . Method according to claim 8, - wherein a second front intermediate signal (Z2v) and a second rear intermediate signal (Z2h) are formed based on the first input signal (E1) and the second input signal (E2) scaled by means of the real-valued amplification factor (m), and
wherein the output signal (out) is generated based on the second overlay (U2) using the adapted second overlay parameter (a2`). Method according to one of the preceding claims - wherein the first rear intermediate signal (Z1h) has a relative attenuation in a frontal direction (6), which is defined in particular based on a direction from the second input converter (M2) to the first input converter (M1), and - wherein the first front intermediate signal (Z1v) has a relative attenuation in a direction opposite to the frontal direction (6). Method according to claim 11, wherein the first front intermediate signal (Z1v) and the first rear intermediate signal (Z1h) are each generated based on a time-delayed superimposition of the two input signals (E1, E2), and whereby the second input signal (E2) is delayed for the first front intermediate signal (Z1v) and the first input signal (E1) is delayed for the first rear intermediate signal (Z1h). Method according to claim 12,
wherein at least in one frequency band a delay is implemented using an all-pass filter Method according to one of the preceding claims, wherein in a first adaptation step a first value of the complex first overlay parameter ( a1 ) is determined, wherein said first value of the first overlay parameter ( a1 ) is converted into the corresponding first and second alternative parameters (ap1, ap2), and the limited second alternative parameter (ap2') is determined from this, wherein a second value of the first overlay parameter (a1) is determined based on the first alternative parameter (ap1) and the limited second alternative parameter ( ap2` ), and wherein said second value of the first overlay parameter ( a1 ) is used for a second adaptation step. Method according to one of the preceding claims,
wherein the first overlay parameter ( a1 ) is determined using a least-mean-squares algorithm and/or using a gradient method. Method according to one of the preceding claims,
wherein the output signal (out) is generated in addition to the superposition of the first and second input signals (E1, E2) using a correction filter (c _cor ) for the frequency response, the correction filter (c _cor ) for the frequency response being selected such that for the Frontal direction (6) the frequency response is flat. Hearing instrument (1), comprehensive - a first input converter (M1) for generating a first input signal (E1) from a sound signal (4) from the environment, - a second input converter (M2) for generating a second input signal (E2) from the sound signal (4) of the environment, and - a control unit (5),
wherein the hearing instrument (1) is set up to carry out the method according to one of the preceding claims.

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