Hearing instrument and method for directional signal processing for a hearing instrument

Abstract

Method for directional signal processing for a hearing instrument (1),- wherein a first input signal (E1) is generated from an ambient sound signal (4) by a first input transducer (M1) of the hearing instrument (1),- wherein a second input transducer (M2) of the hearing instrument (1) is generated from the ambient sound signal (4),- wherein a first anterior intermediate signal (Z1v) and a first posterior intermediate signal (Z1h) are each formed based on the first input signal (E1) and the second input signal (E2),- wherein a first superposition (U1) of the first anterior intermediate signal (Z1v) and the first posterior intermediate signal (Z1h) is formed using a complex-valued first superposition parameter (a1), and is adapted based on the first superposition parameter (a1),- wherein a complex value (a1) resulting from said adaptation of the first superposition (U1)0) of the first superposition parameter (a1) 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 semicircular monotonic relationship to an angle (θ) of minimum sensitivity of the first superposition (U1),- wherein the angle (θ) of minimum sensitivity is modified by a corresponding modification of the second alternative parameter (ap2), thereby forming a modified second alternative parameter, and- wherein an output signal (out) is generated on the basis of 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).

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 of the environment by a first or second input transducer of the hearing instrument, wherein a first front intermediate signal and a first rear intermediate signal are formed in each case on the basis of the first and the second input signal, wherein, in particular on a frequency band basis, a first superposition of the first front intermediate signal and the first rear intermediate signal is formed by means of a first superposition parameter, and is adapted on the basis of the first superposition parameter, and wherein an output signal is generated on the basis of a value of the first superposition parameter and on the basis of a particularly time-delayed superposition of the first input signal and the second input signal.In hearing instruments such as hearing aids for supplying a wearer's hearing loss, a corresponding number of input signals is generated from an ambient sound by a number of input transducers such as microphones, which input signals represent the air pressure fluctuations of the ambient sound at the respective input transducer. On the basis of the input signal or signals, 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 matched to the audiological requirements of the wearer (i.e. for example a hearing weakness), and can in particular include a 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 during the generation of the output signal, which directional signal is directed at an assumed useful signal source (usually a conversation partner or the like), and / or which suppresses interference sources by a spatial "masking out".Such a masking out can be effected by means of a time-delayed superposition of the two input signals, or also by means of two different such overlays, for example by means of a so-called cardioid and an anticardioid signal, which in turn are adaptively superimposed. However, a potential problem here is that the most complete possible masking out of an interference source requires a most identical signal level in the two (or more) input transducers of the hearing instrument. As a result of shading effects both by the head (or also parts of the outer ear) of the wearer and by the housing of the hearing instrument, this is often not the case, which is why an input signal for a complete masking out of a directed interference source is to be adapted accordingly by a then angle-dependent amplification factor. However, such a gain is often difficult to determine. Moreover, such a gain factor can also lead to great fluctuations of a useful signal, which is undesirable. A further challenge is also the additional requirement that the said masking out of the interference source is often to be limited to a specific angle range with respect to the field of view of the wearer (for example to the rear half space).DE 10 2020 210 805 B3 mentions a method for directional signal processing for an acoustic system, wherein a first and a second intermediate signal are generated in each case on the basis of a first and a second input signal of the acoustic system, wherein a preliminary superposition parameter is obtained for a first superposition of the first intermediate signal and the second intermediate signal in such a way that attenuation in a first target direction has a maximum for the first superposition, wherein a superposition parameter is formed on the basis of the preliminary superposition parameter in such a way that a second superposition of the first intermediate signal and the second intermediate signal formed with the superposition parameter has a predetermined first value greater than zero for amplification in the first target direction, and wherein an output signal of the acoustic system is formed on the basis of the second superposition.DE 10 2020 209 555 A1 relates to a method for directional signal processing for a hearing device, wherein a first and a second directional signal are formed on the basis of one and a second input signal of the hearing device, wherein the second directional signal has a relative attenuation in the direction of a first useful signal source, wherein the first directional signal has a relative attenuation in the direction of a second useful signal source, wherein a first gain parameter for a gain of a first useful signal of the first useful signal source and a second gain parameter for a gain of a second useful signal of the second useful signal source are determined, wherein a reference directional characteristic for a reference directional signal is defined, wherein a corrected first and a corrected second gain parameter are determined in such a way on the basis of the first and / or the second gain parameter as a function of the reference directional characteristic, an output directional signal, which is formed as a sum of the first directional signal weighted with the first corrected first gain parameter and the second directional signal weighted with the corrected second gain parameter, transitions into a linearly scaled reference directional signal if the first gain parameter is equal to the second gain parameter.DE 10 2019 205 709 B3 mentions a method for directional signal processing for a hearing device, wherein a first and a second calibration directional signal are generated on the basis of a first and a second input signal of the hearing device, which calibration directional signal each have a relative attenuation in the direction of a first and a second useful signal source of the environment, wherein a relative gain parameter is determined on the basis of the first and the second calibration directional signal, wherein a first and a second processing directional signal are generated on the basis of both the first input signal and the second input signal, wherein a source-sensitive directional signal is generated on the basis of the first and the second processing directional signal and the relative gain parameter, and wherein an output signal of the hearing device is generated on the basis of the source-sensitive directional signal.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 with respect to different signal levels of the individual input signals involved, and which permits an efficient restriction of an angle of the minimum sensitivity of a resultant directional signal.The object mentioned is achieved according to the invention by a method for directional signal processing for a hearing instrument, wherein a first input signal is generated from a sound signal of the environment by a first input transducer of the hearing instrument, wherein a second input signal is generated from the sound signal of the environment by a second input transducer of the hearing instrument, and wherein a first front intermediate signal and a first rear intermediate signal are formed in each case, and preferably by a particularly time-delayed superposition, on the basis of the first input signal and the second input signal.In this case, it is provided that, in particular on a frequency band basis, 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 on the basis of the first superposition parameter, wherein a complex value of the first superposition parameter resulting from said adaptation of the first superposition 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 semi-circle-wise monotone relationship with an angle of minimum sensitivity of the first superposition, wherein the angle of minimum sensitivity is modified via a corresponding modification of the second alternative parameter, and a modified second alternative parameter is formed in the process, and wherein an output signal is generated on the basis of the first alternative parameter and the modified second alternative parameter and on the basis of a superposition of the first input signal and the second input signal. Advantageous and partly per se inventive embodiments are the subject matter of the dependent claims and of the following description.In this case, a hearing instrument generally comprises any device which is configured to generate at least two corresponding input signals by means of at least two input transducers and, on the basis of these, to generate an output signal by means of corresponding processing, which output signal is converted by an output transducer into an output sound signal and is fed to an ear of a wearer of this device. In particular, a headset (e.g. as an "earplug") configured with the corresponding input transducers, a headset, smart glasses with loudspeakers, etc. can be included as a hearing instrument. However, a hearing device in the narrower sense is also included as a hearing instrument, i.e. a device for supplying a hearing weakness of the wearer, in which the input signals generated by means of the input transducers from an ambient sound are processed to form said output signal depending on the audiological requirements of the wearer and in the process are amplified and / or compressed in particular in a frequency band-dependent manner, so that the output sound signal is suitable for at least partially compensating the hearing weakness of the wearer, in particular in a user-specific manner.Included as an (in particular electroacoustic) input transducer is any device which is provided and configured 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 represent the fluctuations of the air pressure of the sound signal and reproduce them within the scope of the respective resolution. In particular, a microphone is included as an input transducer.In particular, the angle of minimum sensitivity is limited as a modification to a predefined angle range via a corresponding limitation of the second alternative parameter, and a limited second alternative parameter is formed as a modified second alternative parameter, wherein an output signal is generated on the basis of the first alternative parameter and the limited second alternative parameter and on the basis of a superposition of the first input signal and the second input signal.On the basis of a modification, in particular a limitation, of the first alternative parameter, it is advantageously possible to modify a minimum sensitivity (that is to say in particular a depth of a so-called "notch") at the corresponding angle.The formation or generation of a resulting signal on the basis of one or more incoming signals is to be understood in particular as meaning that the respective signal components of the incoming signals enter the relevant resulting signal, in particular on a frequency band basis, in accordance with a mapping rule, with the result that there is preferably a monotone, particularly preferably linear relationship between the amplitudes and / or envelopes and / or signal levels of the incoming signals and the respectively corresponding variable of the resulting signal.The first front intermediate signal and the first rear intermediate signal are preferably each generated from the first and the second input signal on the basis of mutually symmetrical mapping rules, in particular as time-delayed superimpositions, so that the directional characteristics of the two first intermediate signals are symmetrical to one another with respect to the free space. However, the first front intermediate signal and the first rear intermediate signal can also be generated on the basis of imaging rules that are different from one another (in particular not symmetrical), wherein the two intermediate signals mentioned are preferably linearly independent of one another. In particular, it is conceivable that one of the intermediate signals has an omnidirectional directional characteristic.The first superposition U 1 is formed in particular in the form wherein Z 1 vand Z 1 heach denote the first front or rear intermediate signal, a 1 ∈ C denotes the first superposition parameter, and ω and t are a frequency index or a discrete time index.The adaptation of the first superposition on the basis of the first superposition parameter comprises, in particular, the fact that the first superposition (the actual superposition, for example according to equation (i), is used here as synonym for the signal resulting from said superposition) is optimized with respect to a characteristic variable such as, for example, the total energy, the total level or a deviation from a reference signal or the like via the first superposition parameter, wherein the optimization can also be carried out numerically in a plurality of steps, such that the first superposition parameter (even for a given time index) converges via the adaptation to a value (which can be determined, for example, on the basis of a limit value for a step width between two adaptation steps).The value of the first superposition parameter, which thus 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 monotone relationship with an angle of minimum sensitivity of the first superposition.This can be motivated in particular on the basis of the following consideration: For a preferably stationary sound signal which impinges on the hearing instrument at an angle of θ with respect to its frontal direction (defined in particular on the basis of the direction from the second to the first input transducer), the relative transfer function from the first to the second input transducer (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 to the first input transducer) is Aθ·e-iωτ cos θ, wherein Aθis an angle-dependent amplitude factor (which takes into account inter alia shadowing effects by the head of the wearer or by the housing of the hearing instrument). With a suitable selection of the two alternative parameters, in particular by a relation between the value of the first superposition parameter and said relative transfer function, an at least semi-circle-wise monotone relation can now be formed between the second alternative parameter and the angle of minimum sensitivity of the first superposition.Under an at least semi-circle-wise monotone relationship, it is in particular comprised 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 one semi-circle, i.e. that a ψ ∈ R exists, so that said monotone relationship applies at least to an angular range of [ψ, ψ + π].On the basis of the second alternative parameter, it is now possible, via the said monotone relationship (and optionally via a sign of the angle and / or a transformation of the angle of minimum sensitivity by π), to limit this angle to a desired, predefined angle range, that is to say, for example, to the rear half space (θ ∈ [90°, 270°]) or a narrower "wedge" in the rear half space (for example, θ ∈ [120°, 240°]), by limiting the value range of the alternative second parameter to a corresponding interval (and, in the process, optionally taking into account a sign of the said angle θ with respect to the frontal or the 180° direction).By said restriction of the angle of minimum sensitivity to the predefined angle range, an adaptation of the alternative second parameter to the interval of the value range of this second alternative parameter corresponding to said angle restriction thus takes place, whereby in particular a generation of the restricted second alternative parameter takes place. 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 predefined angle range, or otherwise be given by a limit value of such an interval.On the basis of this limited second alternative parameter and a superposition of the first and the second input signal, an output signal is now generated. This can be effected in particular by reversing the calculation of the two alternative parameters from the first superposition parameter to the effect that an adapted first superposition parameter is formed on the basis of the first alternative parameter and the limited second alternative parameter, and the superposition of the two input signals for generating the output signal is accordingly given by the first superposition (which, as a result of its generation from the first front and first rear intermediate signal, also represents a superposition of the two input signals), but now with the first adapted superposition parameter.The output signal can be directly converted 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 pass through further signal processing steps (for example further noise suppression and / or frequency band-wise amplification or compression) before the output sound signal is generated therefrom. In particular, a further signal can be added to the output signal before conversion into the output switching signal. Such a pair of alternative parameters can be formed in particular by the fact that the first superposition U1 according to equation (i) is represented by a corresponding conversion on the basis of the two input signals as being associated with the vector of the input signals E T= ( E1, E2) and the coefficient vector w=(w1, w2) T, wherein the coefficients w1and w2depend on the specific form of the generation of the first front and rear intermediate signals Z1v, Z1hin equation (i).In order to gain a better understanding of the coefficient vector w, the underlying adaptation of the first superposition U 1 is assigned to a 1-order finite impulse response (FIR) filter, which then carries out a type of "spatial sampling" of the sound signal. The associated filter polynomial isThe zeros of the polynomial in equation (iii) are z0=-w2 / w1and are unambiguously determined apart from a complex prefactor c ∈C. Accordingly, signals which differ from the first superposition U 1 by such a scalar prefactor c ∈C have the same properties as these with regard to their directivity.In view of the above-mentioned ambiguity of the zero point of the FIR filter according to equation (iii), which is associated with the adaptation of the first superposition, the coefficient vector is now set to w 0= c·[1, - r·e iφ], wherein the relative phase φ and the quotient r of the amounts of the two coefficients w2 / w1, as already mentioned, depend on the specific configuration of the first front and rear intermediate signals Z1v, Z1h. The first alternative parameter can now be formed on the basis of the said quotient r (which therefore specifies the ratio of the amounts of the coefficients), and in particular as this, the second alternative parameter on the basis of the relative phase φ of the coefficients with respect to one another, and in particular as this.This can be seen if, as mentioned above, a stationary sound signal is applied from an angle θ which is to be attenuated as completely as possible. With the above-mentioned relative transfer function between the two input converters, the vector E of the two input signals is obtained as E=E1·h with h=[1, Δθ·e-iωτ cos θ] T. Cancellation of the sound signal then requires in the present illustrationA solution resulting from equation (vi') is r=1 / Δθ, φ= ωt·cos θ. It is clear from this that for θ ∈[0, π], there is a monotone relationship between the relative phase φ of the two coefficients w1, w2of the first superposition (in the illustration of the input signals) and the angle of the minimum sensitivity. This angle can now be limited by the relative phase φ as the second alternative parameter.It has proven to be further advantageous if the value of the first superposition parameter is converted into a corresponding real-valued second superposition parameter and an associated value of a real-valued amplification factor, wherein the real-valued amplification factor is associated with a corresponding gain of the second input signal during the formation of the first front or rear intermediate signal, respectively, and the second superposition parameter is adapted such that for a second superposition, which is formed on the basis of the second superposition parameter from the first front intermediate signal and the first and rear intermediate signal, amplifying the second input signal with said amplification factor, the angle of minimum sensitivity is limited to the predefined angle range, and an adapted second superposition parameter is thereby generated, and that on the basis of the adapted second superposition parameter and the amplification factor and on the basis of an in particular time-delayed superposition of the first input signal and the second input signal are generated, the output signal is generated.The said real amplification factor m ∈R corresponds to an amplification of the second input signal during the formation of the first front and first rear intermediate signal. In other words, the amplification factor m and the second superposition parameter a2 ∈R are determined on the basis of the first superposition parameter a1 ∈C (for the first superposition of the first front and the first rear intermediate signal), i.e. the amplification factor m and the second superposition parameter a2 ∈R, in such a way that the first superposition in this case transitions into a second superposition of the first front and the first rear intermediate signal, wherein in each case the second input signal was amplified, i.e. scaled, beforehand with the amplification factor m in the case of said first intermediate signals, and wherein the second superposition of these intermediate signals is formed on the basis of the second superposition parameter a2 ∈R. In general, this conversion from real and imaginary part from a1 ∈Cto (a2, m) ∈R 2 is well defined.It is not absolutely necessary here for said second superposition (i.e. the resulting signal) to actually be formed (approximately analogously to equation (i)); rather, it is sufficient in this case to carry out only the conversion a1→(a2, m) in accordance with the restrictions (approximately for the amplification factor) which result in particular from the intermediate signals.In the ideal case, the amplification factor is now determined in such a way that a superposition of the first intermediate signals (with a corresponding prior application of the amplification factor to the second input signal during the formation of the intermediate signals) enables a complete suppression of an interference source, and thus takes over the function of a level adaptation between the two input transducers of the hearing instrument. In this case, a monotonic relationship can now generally be established between the second superposition parameter a2and the angle for which the above-mentioned second superposition has a minimum sensitivity or maximum attenuation. Here, in equation (i'), Z2v and Z2h denote second front and second rear intermediate signals, respectively, which are respectively obtained from the first front and first rear intermediate signals by a previous amplification of the second input signal with said amplification factor.As already described, a monotone relationship can now be established between the angle of maximum attenuation of the second superposition U2and the second superposition parameter a2(the monotone is defined here, however, only over angle ranges of half a circle revolution, that is to say for angles θ ∈[γ, γ+π] with γ ∈R).The second superposition parameter can form the second alternative parameter, and the amplification factor can form the first alternative parameter. However, the first alternative parameter can also be formed, as described on the basis of equation (iv'), on the basis of the quotient r of the amounts of the two coefficients w1, w2of the first superposition with respect to the two input signals, and the second alternative parameter on the basis of the relative phase φ of the two coefficients with respect to one another, wherein the second superposition parameter is adapted on the basis of said alternative parameters r and φ, and the adapted second superposition parameter is thus formed.On the basis of this adapted second superposition parameter and a superposition of the first and the second input signal, an output signal is now generated. The said superimposition of the first and the second input signal can be given in particular by the second superimposition (of the second front and second rear intermediate signals) according to equation (i'), wherein the adapted second superimposition parameter a2' (instead of the "original" second superimposition parameter a2) is to be used. In particular, a correction filter for the frequency response can also be added during the generation of the output signal in order to ensure a flat frequency response approximately in the frontal direction (defined, for example, by the direction from the second to the first input transducer of the hearing instrument). In the case where the time delay in the relevant superimpositions is implemented on the basis of a frequency factor, the correction filter can also be given, in particular, by a frequency-dependent correction factor.In particular, therefore, a second front intermediate signal and a second rear intermediate signal are formed on the basis of the first input signal and the second input signal scaled by means of the real-valued amplification factor, preferably by means of an in particular time-delayed superposition, wherein the output signal is generated on the basis of the second superposition using the adapted second superposition parameter.Said superimposition of the two input signals for generating the output signal can, however, also be provided by the first superimposition according to equation (i), wherein, however, the amplification factor m and the adapted second superimposition factor a2' are mapped back again to the then "adapted" value for the first superimposition parameter a1', in particular by means of the reverse mapping rule (a2' m)→a1'.Preferably, the first rear intermediate signal is formed such that it has a relative attenuation in a frontal direction, which is defined in particular on the basis of a direction from the second input transducer to the first input transducer, and the first front intermediate signal is formed such that it has a relative attenuation in a direction opposite to the frontal direction. In particular, the first front and the first rear intermediate signals are symmetrical to each other. In particular, said applies also to the second front and second rear intermediate signals. A relative attenuation is to be understood in particular as a local minimum of sensitivity, preferably a minimum of sensitivity global over all angles. This minimum does not necessarily mean a maximum attenuation in the sense of a total masking out, but can also assume finite values for the respective sensitivity for the first intermediate signals.The first front intermediate signal and the first rear intermediate signal are each advantageously generated on the basis of a time-delayed superposition of the two input signals, wherein the second input signal is delayed for the first front intermediate signal and the first input signal is delayed for the first rear intermediate signal, preferably each by the acoustic propagation time between the two input transducers. As a result, directional signals are generated as first intermediate signals, which have a cardioid or anticardioid-shaped directional characteristic in the free space, and are particularly suitable for the present method as a result of the simple and nevertheless 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 an in particular 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 relevant frequency band. Depending on the implementation, however, this center frequency for the first frequency band may 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. However, this can also be advantageous for other, low frequency bands if the phase within a frequency band has large changes which are mapped only insufficiently with a phase factor which is constant over the relevant frequency band.It has proven to be further advantageous if a first value of the complex first overlay parameter is determined in a first adaptation step, the 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 on the basis of the first alternative parameter and the limited second alternative parameter, and the 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 angle range for the angle of minimum sensitivity of the second superposition does not take place after complete completion of the adaptation of the first superposition parameter. Rather, such a restriction can also take place in a single adaptation step, and the limited second alternative parameter can form the basis for the next adaptation step.The first superposition parameter is advantageously determined by means of a least mean squares algorithm and / or by means of a gradient method. These methods mentioned are particularly suitable for adapting the complex-valued first superposition parameter with real and imaginary parts, i.e. in particular for optimizing the associated first superposition with respect to a characteristic variable via the first superposition parameter. The gradient method can in this case comprise, in particular, an application of a gradient of the real and imaginary part with respect to such a characteristic variable (such as, for example, a signal level or a deviation from an error or reference signal).The invention further specifies a hearing instrument comprising a first input transducer for generating a first input signal from a sound signal of 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 configured to carry out the method described above. The hearing instrument is configured here, in particular by means of the control unit, to carry out the method steps in which in each case one of the input signals or signals derived therefrom is processed. For this purpose, the control unit is equipped, in particular, 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 indicated for the method and for its development can be transferred analogously to the hearing instrument.An exemplary embodiment of the invention is explained in more detail below with reference to drawings. Here, in each case, diagrammatically show: FIG. 1 is a plan view showing directional characteristics of intermediate signals of a hearing instrument, FIG. 2 is a plan view showing the directional characteristics of the intermediate signals of FIG. 1 in the case of unequal signal levels of the input transducers, FIG. 3 is a block diagram of the sequence of a method for directional signal processing in a hearing instrument; and FIG. 4 is a block diagram of an alternative embodiment to the method according to FIG. 3.Mutually corresponding parts and sizes are each provided with the same reference numerals in all figures.FIG. 1 shows schematically in a plan view directional characteristics for a hearing instrument 1. The hearing instrument 1 is configured as a hearing device 2 which is provided and configured to provide a hearing deficiency. The hearing instrument 1 has a first input transducer M 1 and a second input transducer M 2, which are arranged at a distance d from one another and are in the present case each provided by corresponding microphones. A first input signal E 1 is generated from a sound signal 4 of the environment by the first input transducer M 1 and a second input signal E 2 is generated by second input transducer M 2. The hearing instrument 1 further has a control unit 5, which is configured to process said input signals E 1, E 2 and for this purpose comprises in particular a signal processor (not shown in more detail).On the basis of a time-delayed superposition of the first input signal E 1 and the second input signal E 2, a first front intermediate signal Z 1 vis generated, wherein the time delay corresponds exactly to the acoustic propagation time of the distance d:In the ideal case that the signal levels of the first and of the second input signals E 1, E 2 are identical (and in particular no shading effects and no attenuation takes place 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-iωτE1-E2 is generated. In the ideal case already discussed, the first rear intermediate signal Z 1 vhas 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 Z 1 vis opposite to the frontal direction 6.A first superposition U1 according to equation (i) is now formed from the first front and the first rear intermediate signal on the basis of a complex-valued first superposition parameter a1 ∈C, wherein the value of the first superposition parameter a1 (i.e. its real and imaginary part) is determined by an adaptation of the first superposition U1, for example by a minimization of the signal energy or of the level by means of a gradient method. An interference source 8, which contributes a directed interference sound 10 to the sound signal 4 of the environment, can now be "hidden" by means of the first superposition U 1, as the directional characteristic of the first superposition U 1 (solid line) shows. This directional characteristic has the maximum attenuation at the angle θ at which the disturbance source 8 is situated.However, if the signal level for the two input signals E 1, E 2 is not the same, for example as a result of shading effects (for example by the head and / or the pinna of the wearer of the hearing instrument 1, but also by the housing of the hearing instrument 1), then depending on the type of these shading effects, it may happen that, for example, the attenuation for the first rear intermediate signal Z 1 hin the frontal direction 6 is no longer complete, but rather has a finite value. The same may apply to the first front intermediate signal Z 1 vdependently of the specific level differences of the input signals E 1, E 2. As a result, it is possible, if appropriate on the basis of the first superposition U 1 in the direction of the interference source 8, to no longer achieve complete attenuation and therefore also no longer complete masking of the interference sound 10.This situation is schematically illustrated in a plan view in FIG. 2. The first front intermediate signal Z1v (dashed line) and the first rear intermediate signal Z1h now each have a directional characteristic which does not allow a complete more attenuation in some directions. For this reason, the first superposition U 1 (not shown), formed according to equation (i) on the basis of the first front and rear intermediate signals Z 1 v, Z 1 hin accordance with FIG. 2, is also not suitable in the present case for completely blocking out the interference source 8.To solve this problem, a method is proposed which is illustrated by means of a block diagram in FIG. 3. In FIG. 3, the sound signal 4 of the environment according to FIG. 1, which includes the interference sound 10 of the directed interference source 8 (in each case not shown), is converted by the first or second input transducer M 1, M 2 into the first or second input signal E 1, E 2. The first front or first rear intermediate signal Z1v, Z1h is formed from the two input signals E1, E2 in each case by time-delayed superposition (see description of FIG. 1, in particular equation (ii')):Generally, the signal levels of the first and second input signals E1, E2 are not equal, so that the first front and first rear intermediate signals Z1v, Z1h, respectively, have directional characteristics comparable to those shown in FIG. 2.On the basis of a complex first superposition parameter a1 ∈C, a first superposition U1according to equation (i) is now formed from the first front and the first rear intermediate signals Z1v, Z1h. This first superposition U 1 is subjected to an adaptation 12, in which a specific value a1.0 for the first superposition parameter a 1 is ascertained. The adaptation 12 can be effected, for example, in a minimization of the signal energy of the first superposition U 1 by a gradient method with respect to the real and imaginary part of the first superposition parameter a 1, or the like.According to equation (i), (v") and (vi), for the first superposition U1: with the vector of the input signals E T= ( E1, E2) and the coefficient vector w=(w1, w2) T, the specific form of the coefficients w1and w2being given by equation (vii) now.For the further procedure, therefore, the coefficient vector w according to equation (vii) is converted into the form w 0 according to equation (iv'), resulting. The relative phase φ is here simply obtained from the argument of 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 eiφ-iωτ cosθ resulting from equation (iv') in order to limit an angle range Δθ for the angle θ, resulting in a limited relative phase φ' or a limited alternative second parameter ap2'. 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 superposition U1is already in the desired angular range Δθ (approximately the rear half space with respect to the frontal direction 6).Due to the adapted relative phase φ', in equation (viii), a corresponding adaptation of the first superposition parameter a1 also takes place. The equation (viii) can then be resolved according to this adapted first superposition parameter a1' toOn the basis of the first superposition U1' (dashed-dotted line) thus adjusted with said adjusted first superposition parameter a1', an output signal Out can now be generated, wherein the adjusted first superposition U1' is, among other things, in particular still multiplied by a correction factor c cor for correcting the frequency response, so that the frequency response of the output signal Out is flat in the frontal direction 6. Furthermore, further signal processing steps 20 such as noise or feedback suppression etc., but also frequency band-dependent increase depending on the audiological requirements of the carrier or the like, can be interposed.FIG. 4 shows an alternative embodiment of the method according to FIG. 3 on the basis of a block diagram. As in that, the first superposition U1is formed on the basis of the first superposition parameter a1, and the value a1.0of the first superposition parameter a1is determined in the adaptation 12. In a next step, the value a1.0 of the first superposition parameter a1 is now mapped onto a real-valued second superposition parameter a2 ∈R and a real-valued amplification factor m ∈R, the latter being assigned to the second input signal E2.In order to be able to determine the relationship between the first superposition parameter a1 (or from its value a1.0) and the concrete values of the second superposition parameter a2 and the amplification factor m, a second front intermediate signal Z2v and a second rear intermediate signal Z2h are defined (dashed signal path), in which, however, the second input signal E2 is respectively acted upon by the amplification factor m, that is to sayThis amplification factor m can be used to compensate for a different signal level between the first and the second input signal E 1, E 2. Therefore, even in the case of different signal levels, the second front and second rear intermediate signals Z2v, Z2h have the directional characteristics shown in FIG. 1, which are no longer applicable to the first front and first rear intermediate signals Z1v, Z1h in the general case (i.e. not in free space, but in shading effects etc.) (for this general case, these directional signals have directional characteristics according to FIG. 2 as described).If a second superposition U2 (dashed signal path) analogous to equation (i) is now formed 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, respectively, by said amplification factor m in the proportion of the second input signal E2) on the basis of the second superposition parameter a2.The amplification factor m and the second superposition parameter a 2 are to be determined here in such a way that the illustration is intended to enable the angle θ of the maximum attenuation (see FIG. 1 ) to be restricted to a desired angular range.In a manner analogous to equations (vii') and (viii), the relationship between the relative phase φ and the coefficient quotient r=|w2' / w1'| on the one hand and the amplification factor m as the first alternative parameter ap1and the second superposition parameter a2 as the second alternative parameter ap2on the other hand is now produced from equation (x), using the consideration postulated from equation (iii) and (iv').In this case, use was made of the fact that the zeros of the polynomial in equation (iii) are defined only up to a factor c ∈C, as a result of which w2 / w1=w2' / w1' follows. Since the fraction on the right side is of the magnitude 1, the amplification factor m=r is obtained, wherein r is given by the magnitude of equation (viii) on the basis of the value a1.0 of the first superposition parameter a1.For the second superposition parameter a2as second alternative parameter ap2of the method according to FIG. 4, equation (xi) results:On the basis of corresponding tabulated values, it is now possible to produce via 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 (eiφ -iωτ cos θ=1; see equation (iv')) and thus also of the second superposition U2 (which initially represents only a conversion of the first superposition U1).From this, a corresponding adjusted value for the second superposition parameter a2, i.e. an adjusted second superposition parameter a2' or a limited second alternative parameter ap2' can be determined.The output signal out can now be formed (possibly 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 signal Z2v, Z2h according to equation (ix), but on the basis of the adapted second superposition parameter a2' (instead, as in equation (i'), on the basis of the second superposition parameter a2). The amplification factor m in the second front and second rear intermediate signals Z2v, Z2h according to equation (ix) is thereby evident as r=m according to equation (xi), where r according to equation (viii) from the first superposition parameter a1.However, the amplification factor m and the adapted second superposition parameter a2' can also be back-calculated into the domain of the first superposition parameter a1 (not shown), so that the output signal out is then formed in that case from a first superposition on the basis of the thus determined adapted first superposition parameter a1'. 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 superposition U 1, is independent of the angle θ of the minimum sensitivity.Although the invention has been illustrated and described in more detail by the preferred exemplary embodiment, the invention is not restricted by the disclosed examples and other variations can be derived therefrom by the person skilled in the art without departing from the scope of protection of the invention.List of reference characters1 Hearing instrument 2 Hearing device 4 Sound signal (of the environment) 5 Control unit 6 Frontal direction 8 Interference source 10 Interference sound 12 Adaptation 20 Signal processing steps a1(') (adapted) First superposition parameter a2(') (adapted) Second superposition parameter ap1, ap2First and second alternative parameters ap2' Limited Second alternative parameter c cor Correction factor E1, E2First and second input signal M1, M2First and second input converter out Output signal m Amplification factor r Quotient (of the amounts of the coefficients) U1, U2First and second superpositions w1('), w2(') Coefficients Z1v, Z1hFirst front and first rear intermediate signal Z2v, respectively, Z2h second front and second rear intermediate signals Δθ angle range θ angle (minimum sensitivity) τ time delay φ relative phase (coefficient)

Claims

Method for directional signal processing for a hearing instrument (1), - wherein a first input signal (E1) is generated from a sound signal (4) of 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), - wherein a first front intermediate signal (Z1v) and a first rear intermediate signal (Z1h) are each formed on the basis of the first input signal (E1) and the second input signal (E2), - wherein a first superposition (U1) of the first front intermediate signal (Z1v) and of 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), - wherein a complex value (a1.0) of the first superposition parameter (a1) resulting from said adaptation of the first superposition (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 semi-circle-wise monotone relationship with an angle (θ) of minimum sensitivity of the first superposition (U1), - wherein the angle (θ) of minimum sensitivity is modified via a corresponding modification of the second alternative parameter (ap2), and a modified second alternative parameter is formed in the process, and - wherein an output signal (out) is generated on the basis of the first alternative parameter (ap1) and the modified second alternative parameter and 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 formation of the first superposition (U1) of the first front intermediate signal (Z1v) and of the first rear intermediate signal (Z1h) takes place by means of the complex-valued first superposition parameter (a1) and the adaptation takes place on the basis of the first superposition parameter (a1) on a frequency band-by-frequency basis.Method according to Claim 1 or Claim 2, wherein the angle (θ) of minimum sensitivity is limited as a modification to a predefined angle range (Δθ) by means of 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 - wherein an output signal (out) is generated on the basis of the first alternative parameter (ap1) and the limited second alternative parameter (ap2') and on the basis of a superposition of the first input signal (E1) and the second input signal (E2).Method according to one of the preceding claims, wherein a minimum sensitivity at the corresponding angle (θ) is modified on the basis of a modification of the first alternative parameter (ap1).The method of claim 4, wherein the minimum sensitivity at the corresponding angle (θ) is modified based on 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 of the second input signal (E2) is formed in the first superposition (U1), wherein the first alternative parameter (ap1) is formed on the basis of a quotient (r) of the amounts of the two coefficients (w1, w2), and wherein the second alternative parameter (ap2) is formed on the basis of a relative phase (φ) of the two coefficients (w1, w2) to one another.Method according to claim 5, wherein an adapted first superposition parameter (a1') is formed on the basis of the first alternative parameter (ap1) and the limited second alternative parameter (ap2'), and wherein the superposition for generating the output signal (out) is formed by the first superposition (U1) on the basis of the adapted first superposition parameter (a1').Method according to one of the preceding claims, - wherein the value of the first superposition parameter (a1) is converted into a corresponding real-valued second superposition parameter (a2) and an associated value of a real-valued amplification factor (m), wherein the real-valued amplification factor (m) is associated with a corresponding gain of the second input signal (E2) during the formation of the first front and / or rear intermediate signal (Z1v, Z1h), - wherein the second superposition parameter (a2) is adapted such that, for a second superposition (U2) which is formed on the basis of the second superposition parameter (a2), from the first front intermediate signal (Z1v) and the first rear intermediate signal (Z1h) while amplifying the second input signal (E2) with said amplification factor (m), the angle (θ) of minimum sensitivity is limited to the predetermined angle range (Δθ), and an adapted second superposition parameter (a2') is thereby generated, and - wherein the output signal (out) is generated on the basis of the adapted second superposition parameter (a2') and the gain factor (m) and on the basis of a superposition of the first input signal (E1) and the second input signal (E2).Method according to claim 7, wherein 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 8 in conjunction with Claim 6, wherein the first alternative parameter (ap1) is formed on the basis of the quotient (r) of the amounts of the two coefficients (w1, w2) of the first input signal (E1) and of the second input signal (E2) in the first superposition (U1), wherein the second alternative parameter (ap2) is formed on the basis of the relative phase (φ) of the two coefficients (w1, w2) with respect to one another, and wherein the second superposition parameter (a2) is matched on the basis of the first alternative parameter (ap1) and the limited second alternative parameter (ap2'), and the matched second superposition parameter (a2') is thus formed.Method according to one of Claims 8 to 10, wherein the output signal (out) is generated on the basis of the first superposition (U1), wherein for this purpose the value (a1.0) of the adapted first superposition parameter (a1) is determined on the basis of the adapted second superposition parameter (a2') and on the basis of the amplification factor (m).Method according to Claim 10, - wherein a second front intermediate signal (Z2v) and a second rear intermediate signal (Z2h) are formed in each case on the basis of 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 on the basis of the second superposition (U2) using the adapted second superposition 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), 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 13, wherein the frontal direction (6) is defined by a direction from the second input transducer (M2) to the first input transducer (M1).Method according to Claim 13 or Claim 14, wherein the first front intermediate signal (Z1v) and the first rear intermediate signal (Z1h) are each generated on the basis of a time-delayed superposition of the two input signals (E1, E2), and wherein 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 15, wherein a delay is implemented by means of an all-pass filter at least in one frequency bandMethod according to one of the preceding claims, wherein in a first adaptation step a first value of the complex first superposition parameter (a1) is determined, wherein said first value of the first superposition parameter (a1) is converted into the corresponding first and second alternative parameters (ap1, ap2), and from this the limited second alternative parameter (ap2') is determined, wherein a second value of the first superposition parameter (a1) is determined on the basis of the first alternative parameter (ap1) and the limited second alternative parameter (ap2'), and wherein said second value of the first superposition parameter (a1) is used for a second adaptation step.Method according to one of the preceding claims, wherein the first superposition parameter (a1) is determined by means of a least mean squares algorithm and / or by means of 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 the second input signal (E1, E2), on the basis of a correction filter (c cor) for the frequency response.Method according to claim 19, wherein the frequency response correction filter (c cor) is chosen such that the frequency response is flat for the frontal direction (6).Hearing instrument (1) comprising - a first input transducer (M1) for generating a first input signal (E1) from a sound signal (4) of the environment, - a second input transducer (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 configured to carry out the method according to one of the preceding claims.

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