METHOD FOR DIRECTIONAL SIGNAL PROCESSING FOR A HEARING INSTRUMENT
Patent Information
- Application Number
- AT2023179264T
- Authority / Receiving Office
- AT · AT
- Patent Type
- Patents
- Current Assignee / Owner
- Priority Date
- 2022-07-06
- Filing Date
- 2023-06-14
- Publication Date
- 2026-04-15
- Estimated Expiration
- 2043-06-14
Abstract
Description
[0001] The invention relates to a method for directional signal processing for a hearing instrument, wherein a first and a second input signal are generated from a sound signal of the environment by a first and a second input transducer of the hearing instrument, wherein a first front intermediate signal and a first rear intermediate signal are formed on the basis of the first and the second input signal, wherein, in particular frequency band by frequency, 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 superposition, in particular with a time delay, of the first input signal and the second input signal.
[0002] In hearing instruments such as hearing aids used to treat a wearer's hearing loss, a number of input transducers, such as microphones, generate a corresponding number of input signals from ambient sound. These input signals represent the air pressure fluctuations of the ambient sound at the respective input transducer. Based on the input signal(s), an output signal is generated by signal processing, which is then 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 wearer's audiological requirements (e.g., hearing impairment) and, in particular, can include frequency band-specific amplification and / or compression.
[0003] 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 be performed. This allows a directional signal to be generated, for example, as an intermediate signal during output signal generation. This signal is directed toward a presumed desired signal source (usually a conversation partner or similar) and / or suppresses interference sources by spatially "masking" them.
[0004] Such masking can be achieved by means of a time-delayed superposition of the two input signals, or by means of two different such superpositions, 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 the most complete masking of an interference source requires a signal level that is as identical as possible in the two (or more) input transducers of the hearing instrument. Due to shadowing effects both by the wearer's head (or parts of the outer ear) and by the housing of the hearing instrument, this is often not possible, which is why an input signal for complete masking of a directed interference source must be adjusted accordingly using an angle-dependent gain factor. However, such a gain factor is often difficult to determine.Furthermore, such a gain factor can lead to significant fluctuations in the desired signal, which is undesirable. A further challenge is the additional requirement that the suppression of the interference source often has to be limited to a specific angular range within the wearer's field of vision (e.g., the rear half-space).
[0005] 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.
[0006] The stated object 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 on the basis of the first input signal and the second input signal, and preferably by a particularly time-delayed superposition.
[0007] It is provided 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, 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 semicircular monotonic relationship to an angle of minimum sensitivity of the first superposition, wherein the angle of minimum sensitivity is modified by a corresponding modification of the second alternative parameter, and in the process a modified second alternative parameter is formed,and wherein an output signal is generated based on the first alternative parameter and the modified second alternative parameter, as well as on the basis of a superposition of the first input signal and the second input signal. Advantageous and, in part, inventive embodiments are the subject of the dependent claims and the following description.
[0008] A hearing instrument generally includes any device configured to generate at least two corresponding input signals using at least two input transducers. Based on these signals, an output signal is generated through appropriate processing. This output signal is converted by an output transducer into an output sound signal and delivered to the ear of a wearer of this device. In particular, a hearing instrument can include headphones equipped with the corresponding input transducers (e.g., as an "earplug"), a headset, data glasses with a loudspeaker, 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 the said output signal depending on the audiological requirements of the wearer and are amplified and / or compressed in particular depending on the frequency band, so that the output sound signal is suitable, in particular in a user-specific manner, to at least partially compensate for the hearing impairment of the wearer.
[0009] An input transducer (particularly an electroacoustic one) includes any device intended and configured to generate a corresponding electrical signal (the associated input signal) from the ambient sound signal, the voltage or current fluctuations of which preferably represent the fluctuations in the air pressure of the sound signal and reproduce them within the respective resolution. In particular, a microphone is included as an input transducer.
[0010] 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, wherein an output signal is generated based on the first alternative parameter and the limited second alternative parameter and based on a superposition of the first input signal and the second input signal.
[0011] By modifying, in particular limiting, the first alternative parameter, a minimum sensitivity (i.e. in particular a depth of a so-called "notch") at the corresponding angle can be advantageously modified.
[0012] 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 by frequency, are incorporated into the relevant resulting signal according to a mapping rule, so that preferably a monotonic, particularly preferably linear relationship exists between the amplitudes and / or envelopes and / or signal levels of the incoming signals and the respective corresponding size of the resulting signal.
[0013] The first front intermediate signal and the first rear intermediate signal are preferably each generated from the first and second input signals using mutually symmetrical mapping rules, in particular as time-delayed superpositions, so that the directional characteristics of the two first intermediate signals are symmetrical to one another with respect to free space. However, the first front intermediate signal and the first rear intermediate signal can also be generated using different (in particular non-symmetrical) mapping rules, wherein the two intermediate signals are preferably linearly independent of one another. In particular, it is conceivable for one of the intermediate signals to have an omnidirectional directional characteristic.
[0014] The first overlay U1 is formed in particular in the form U 1 ω , t = Z 1 v ω , t + a 1 _ ω , t ⋅ Z 1 h ω , t , where Z1v and Z1h denote the first front and rear intermediate signals, respectively, a 1 _ ∈ ℂ the first overlay parameter, and ω and t are a frequency and a discrete time index, respectively.
[0015] The adaptation of the first superposition based on the first superposition parameter comprises in particular that the first superposition (the actual superposition, e.g. according to equation (i), is used synonymously with the signal resulting from the said superposition) is optimized with respect to a characteristic such as 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 several steps, so that the first superposition parameter (even for a given time index) converges via the adaptation to a value (which can be determined, for example, based on a limit value for a step size between two adaptation steps).
[0016] The value of the first overlay parameter, which generally has a real and an 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 of the first overlay.
[0017] 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 by 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 (i.e. the amplitude and phase difference due to the propagation of the sound signal from the sound source at the angle θ to the second instead of the first input transducer) is A θ · e -iωτ cos θ< , where A θ is an angle-dependent amplitude factor (which takes into account, among other things, shadowing effects by the wearer's head or by the housing of the hearing instrument).With a suitable choice of the two alternative parameters, in particular by a relation between the value of the first superposition parameter and the said relative transfer function, an at least semicircular monotonic relationship can be formed between the second alternative parameter and the angle of minimum sensitivity of the first superposition.
[0018] An at least semicircle-wise monotonic relationship includes in particular that the relationship between the second alternative parameter and the angle of minimum sensitivity applies at least for an angular range of the said angle which covers at least one semicircle, ie that a ψ ∈ ℝ exists, so that the said monotonic relationship holds at least for an angular range of [ψ, ψ + π].
[0019] Using the second alternative parameter, this angle can now be limited to a desired, predetermined angular range, e.g., to the rear half-space (θ ∈ [90°, 270°]), or a narrower "wedge" in the rear half-space (e.g., θ ∈ [120°, 240°]), by using the aforementioned monotonic relationship (and, if necessary, a sign of the angle and / or a transformation of the angle of minimum sensitivity by π), by restricting the value range of the alternative second parameter to a corresponding interval (and, if necessary, taking into account a sign of the said angle θ with respect to the frontal or 180° direction).
[0020] By limiting the angle of minimum sensitivity to the specified angular range, the alternative second parameter is adapted to the interval of the value range of this second alternative parameter corresponding to the said angular restriction, thereby generating, in particular, the limited second alternative parameter. This limited second alternative parameter can preferably be identical to the second alternative parameter if the associated angle θ of minimum sensitivity already lies within the specified angular range, or otherwise be specified by a limit value of such an interval.
[0021] Based on this limited second alternative parameter and a superposition of the first and second input signals, an output signal is now generated. This can be achieved, in particular, by reversing the calculation of the two alternative parameters from the first superposition parameter, such that an adjusted first superposition parameter is formed based on the first alternative parameter and the limited second alternative parameter. Accordingly, the superposition of the two input signals to generate the output signal is determined by the first superposition (which, due to its generation from the first front and first rear intermediate signals, also represents a superposition of the two input signals), but now with the first adjusted superposition parameter.
[0022] 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 then fed to the hearing of the hearing instrument wearer. Alternatively, the output signal of the method can undergo further signal processing steps (such as further noise suppression and / or frequency band-wise amplification or compression) before the output sound signal is generated. In particular, a further signal can be added to the output signal before conversion into the output sound 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 U 1 = E 1 ⋅ w 1 + E 2 ⋅ w 2 = E T ⋅ 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 concrete form of generation of the first front and rear intermediate signals Z1v, Z1h in equation (i).
[0023] To better understand the coefficient vector w To obtain the desired signal, the underlying adaptation of the first overlay U1 is assigned to a first-order finite impulse response (FIR) filter, which then performs a kind of spatial sampling of the sound signal. The corresponding filter polynomial is P z = w 1 + w 2 ⋅ z − 1 .
[0024] The zeros of the polynomial in equation (iii) are z0 = - w2 / w1, and are, up to a complex prefactor, c _ ∈ ℂ uniquely determined. Accordingly, signals that differ from the first superposition U1 by such a scalar prefactor c _ ∈ ℂ differ, have the same properties in terms of their directivity as these.
[0025] Given the aforementioned ambiguity of the zero 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 magnitudes 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 using the aforementioned quotient r (which thus indicates the ratio of the magnitudes of the coefficients), and in particular as this, the second alternative parameter using the relative phase ϕ of the coefficients to each other, and in particular as this.
[0026] 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. Using the relative transfer function between the two input transducers mentioned above, the vector E of the two input signals as E = E1 · h with h = [1, A θ · e -iωτ cos θ< ] T< . An extinction of the sound signal then requires in the present representation h T ⋅ w 0 = 0 , or 1 , A θ ⋅ e − i ωτ cos θ ⋅ 1 , − r ⋅ e iϕ T = 0 .
[0027] A solution resulting from equation (vi`) is r = 1 / A θ , ϕ = ωt · cos θ . This clearly shows 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 minimum sensitivity. This angle can now be limited using the relative phase ϕ as a second alternative parameter.
[0028] It is 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 corresponds to a corresponding amplification of the second input signal in the formation of the first front or rear input signal.rear intermediate signal is assigned, 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 signals with amplification of the second input signal with the said amplification factor, the angle of minimum sensitivity is limited to the predetermined angular range, and in this way an adapted second superposition parameter is generated, and that the output signal is generated on the basis of the adapted second superposition parameter and the amplification factor and on the basis of a particularly time-delayed superposition of the first input signal and the second input signal.
[0029] The said real gain factor m ∈ 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 superposition parameter a 1 _ ∈ 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 a 2 ∈ ℝ determined in such a way that the first superposition merges into a second superposition of the first front and the first rear intermediate signal, wherein in said first intermediate signals the second input signal was previously amplified, i.e. scaled, with the amplification factor m, and wherein the second superposition of these intermediate signals is formed on the basis of the second superposition parameter a 2 ∈ ℝ In general, this conversion of real and imaginary parts of a 1 _ ∈ ℂ after a 2 , m ∈ ℝ 2 well-defined.
[0030] It is not absolutely necessary that the second superposition (i.e. the resulting signal) is actually formed (e.g. analogous to equation (i)); it is sufficient to simply convert a1 → (a2, m) according to the restrictions resulting in particular from the intermediate signals (e.g. for the gain factor).
[0031] Ideally, the amplification factor is determined in such a way that a superposition of the first intermediate signals (with 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, thus assuming the function of level adjustment between the two input transducers of the hearing instrument. In this case, a monotonic relationship can generally be established between the second superposition parameter a2 and the angle for which the above-mentioned second superposition U 2 ω , t = Z 2 v ω , t + a 2 ω , t ⋅ Z 2 h ω , t , has a minimum sensitivity and maximum attenuation, respectively. In equation (i'), Z2v and Z2h denote a second front and second rear intermediate signal, respectively, which are derived from the first front and first rear intermediate signal, respectively, by a prior amplification of the second input signal with the said amplification factor.
[0032] As already described, a monotonic relationship can now be established between the angle of maximum attenuation of the second superposition U2 and the second superposition parameter a2 (the monotonicity is, however, only defined over angular ranges of half a circle, i.e. for angles θ ∈ [γ, γ + π ] with γ ∈ ℝ ).
[0033] The second superposition parameter can form the second alternative parameter, and the gain factor the first alternative parameter. However, the first alternative parameter can also be formed, as described in equation (iv`), based on the quotient r of the magnitudes 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, whereby the adaptation of the second
[0034] superposition parameter is determined using the said alternative parameters r and ϕ, and thus the adapted second superposition parameter is formed.
[0035] Based on this adjusted second superposition parameter and a superposition of the first and second input signals, an output signal is then generated. Said superposition of the first and second input signals can be achieved, in particular, by the second superposition (of the second front and second rear intermediate signals) according to equation (i'), whereby the adjusted second superposition parameter a2' (instead of the "original" second superposition parameter a2) is used. When generating the output signal, a correction filter for the frequency response can be added, in particular, 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 case where the time delay in the relevant superpositions is implemented using a frequency factor, the correction filter can in particular also be provided by a frequency-dependent correction factor.
[0036] In particular, 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 a particularly time-delayed superposition, wherein the output signal is generated on the basis of the second superposition using the adapted second superposition parameter.
[0037] 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), whereby, however, the gain factor m and the adjusted second superposition factor a2' are again adjusted back to the then "adjusted" value for the first superposition parameter a1 ' is mapped, in particular by means of the reverse mapping rule (a2', m) → a1' .
[0038] Preferably, the first rear intermediate signal is formed such that it has a relative attenuation in a frontal direction, which is defined in particular by 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 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 the sensitivity across all angles. This minimum does not necessarily have to mean a maximum attenuation in the sense of total blanking, but can in particular also assume finite values for the respective sensitivity for the first intermediate signals.
[0039] Advantageously, the first front intermediate signal and the first rear intermediate signal are each generated by a time-delayed superposition of the two input signals, with the second input signal being delayed for the first front intermediate signal and the first input signal being delayed for the first rear intermediate signal, preferably by the acoustic propagation time between the two input transducers. This generates directional signals as the first intermediate signals, which have a cardioid or anticardioid directional characteristic in free space and are particularly suitable for the present method due to their simple yet stable generation.
[0040] Expediently, a delay between the input signals, in particular in the time-frequency domain, is implemented by means of an additional all-pass filter, in particular in at least one frequency band, preferably up to a band cut-off frequency of up to 500 Hz. In the time-frequency domain, a delay can be implemented using a phase factor which depends on the center frequency of the relevant frequency band. Depending on the implementation, however, this center frequency can be 0 Hz for the first frequency band, so that no delay would be possible. In this case, an alternative implementation of the delay using an all-pass filter is advantageous. However, this can also be advantageous for other, lower frequency bands if the phase within a frequency band exhibits large changes which are only inadequately represented by a phase factor that is constant across the relevant frequency band.
[0041] 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 from this, the limited second alternative parameter is determined, 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 for the restriction of the angular range for the angle of the minimum sensitivity of the second overlay to occur after the complete completion of the adaptation of the first overlay parameter.Rather, such a restriction can also be made in a single adaptation step, and the limited second alternative parameter can form the basis for the next adaptation step.
[0042] Advantageously, the first overlay parameter is determined using a least-mean-squares algorithm and / or a gradient method. These methods are particularly suitable for adapting the complex-valued first overlay parameter with real and imaginary components, i.e., in particular, for optimizing the associated first overlay with respect to a characteristic via the first overlay parameter. The gradient method can, in particular, comprise applying a gradient of the real and imaginary components with respect to such a characteristic (such as a signal level or a deviation from an error or reference signal).
[0043] The invention further relates to a hearing instrument comprising a first input transducer for generating a first input signal from an ambient sound signal, a second input transducer for generating a second input signal from the ambient sound signal, and a control unit, wherein the hearing instrument is configured to carry out the method described above. The hearing instrument is configured, 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 equipped, in particular, with at least one signal processor.
[0044] 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 applied analogously to the hearing instrument.
[0045] An embodiment of the invention is explained in more detail below with reference to the accompanying drawings, each of which shows schematically: Fig. 1 shows a plan view of the directional characteristics of intermediate signals of a hearing instrument, Fig. 2 shows a plan view of the directional characteristics of the intermediate signals according to Fig. 1 in the case of unequal signal levels of the input transducers, 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 a method according to Fig. 3 alternative design.
[0046] Corresponding parts and sizes are provided with the same reference numerals in all figures.
[0047] In Figure 1is shown schematically in a plan view of directional characteristics for a hearing instrument 1. The hearing instrument 1 is designed as a hearing aid 2, which is intended and configured to treat a hearing impairment. The hearing instrument 1 has a first input transducer M1 and a second
[0048] Input transducers M2, which are arranged at a distance d from one another and, in this case, are each provided by corresponding microphones. From a sound signal 4 from the environment, a first input signal E1 is generated 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 configured to process said input signals E1, E2 and, for this purpose, in particular comprises a signal processor (not shown in detail).
[0049] By means of a time-delayed superposition of the first input signal E1 and the second input signal E2, a first front intermediate signal Z1v is generated, whereby the time delay corresponds exactly to the acoustic propagation time of the distance d: Z 1 v ω , t = E 1 ω , t − E 2 ω , t − τ , or Z 1 v ω , t = E 1 ω , t = e − i ωτ E 2 ω , t
[0050] 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 shadowing effects and no attenuation occurs 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 a 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.
[0051] From the first front and the first rear intermediate signal, a complex-valued first superposition parameter is now used to a 1 _ ∈ ℂ a first overlay U1 is formed according to equation (i), wherein the value of the first overlay parameter a1 (i.e., its real and imaginary parts) can be determined by adapting the first superposition U1, for example, by minimizing the signal energy or level using a gradient method. An interference source 8, which contributes a directional noise 10 to the ambient sound signal 4, can now be "masked" by the first superposition U1, as shown by the directional characteristic of the first superposition U1 (solid line). This directional characteristic exhibits maximum attenuation at the angle θ, at which the interference source 8 is located.
[0052] However, if the signal level for the two input signals E1, E2 is not the same, for example due to shadowing effects (e.g. by the head and / or pinna of the wearer of the hearing instrument 1, but also by the housing of the hearing instrument 1), then depending on the nature of these shadowing effects it may happen 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 similar situation may apply to the first front intermediate signal Z1v, depending on the specific level differences of the input signals E1, E2. As a result, it may no longer be possible to achieve complete attenuation and thus complete suppression of the background noise 10 using the first superposition U1 in the direction of the interference source 8.
[0053] This situation is shown schematically in a top view in Figure 2The first front intermediate signal Z1v (dashed line) and the first rear intermediate signal Z1h now each have a directional characteristic that no longer allows complete attenuation in some directions. For this reason, the first superposition U1 (not shown), formed according to equation (i) from the first front and rear intermediate signals Z1v, Z1h, is Figure 2 , in the present case not suitable for completely suppressing the interference source 8.
[0054] To solve this problem, a method is proposed which is based on a block diagram in Figure 3 is shown. In Figure 3 the sound signal 4 of the environment is Figure 1, which includes the noise 10 of the directed noise source 8 (not shown in each case), is converted by the first and second input transducers M1, M2 into the first and second input signals E1, E2, respectively. From the two input signals E1, E2, the first front and first rear intermediate signals Z1v, Z1h are formed by time-delayed superposition (see description of Figure 1 , in particular equation (ii`)): Z 1 v = E 1 − e − i ωτ E 2 , Z 1 h = e − i ωτ E 1 − E 2 .
[0055] 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 shown.
[0056] Using a complex first overlay parameter a 1 _ ∈ ℂ A first superposition U1 is now formed from the first front and the first rear intermediate signal Z1v, Z1h according to equation (i). This first superposition 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 be carried out, for example, 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.
[0057] According to equations (i), (v") and (vi), the first superposition U1 is: U 1 = E 1 ⋅ 1 + a 1 _ e − i ωτ − E 2 ⋅ e − i ωτ + a 1 _ = E 1 ⋅ w 1 + E 2 ⋅ w 2 , U 1 = E ⊤ ⋅ 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).
[0058] For the further procedure, the coefficient vector w according to equation (vii) in the formw 0 according to equation (iv'), where r e i φ = e − i ωτ + a 1 _ 1 + a 1 _ e − i ωτ 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 magnitudes 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, 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 U1 already lies in the desired angular range Δθ (e.g. the rear half-space with respect to the frontal direction 6).
[0059] Due to the adjusted relative phase ϕ', a corresponding adjustment of the first superposition parameter is also carried out in equation (viii) a1. Equation (viii) can then be adapted to this first overlay parameter a1 ' be resolved to a 1 ′ _ = e − i ωτ − r e i φ ′ r e i φ ′ - i ωτ − 1 .
[0060] Based on the thus adjusted first overlay U1' (dashed-dotted line) with said adjusted first overlay parameter a1 ', an output signal Out can now be generated, wherein the adjusted first superposition U1' is multiplied, among other things, by a correction factor c cor to correct 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 boosting depending on the wearer's audiological specifications, or similar, can be interposed.
[0061] In Figure 4 An alternative embodiment of the method according to Figure 3 As in that case, the first overlay U1 is formed based on the first overlay parameter a1, and in the adaptation 12 the value a1 .0 of the first overlay parameter a1 In the next step, the value a1 .0 of the first overlay parameter a1 to a real-valued second overlay parameter a 2 ∈ ℝ and a real-valued gain factor m ∈ ℝ , the latter being assigned to the second input signal E2.
[0062] To determine the relationship between the first overlay parameter a1 (or from its value a1.0) and the concrete values of the second superposition parameter a2 and the gain 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 each subjected to the gain factor m, i.e. Z 2 v = E 1 − m ⋅ e − i ωτ E 2 , Z 2 h = e − i ωτ E 1 − m ⋅ E 2 .
[0063] This amplification factor m can compensate for a different signal level between the first and second input signals E1, E2. Therefore, even in the case of different signal levels, the second front and second rear intermediate signals Z2v, Z2h have the Figure 1shown directional characteristics, which are no longer valid for the first front and first rear intermediate signal Z1v, Z1h in the general case (i.e. not in free space, but in the case of shadowing effects etc.) (for this general case, these directional signals have directional characteristics as described according to Figure 2 on).
[0064] If one now forms a second superposition U2 (dashed signal path) analogous to equation (i) from the 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 superposition parameter a2, the following applies: U 2 = E 1 ⋅ 1 + a 2 e − i ωτ − m ⋅ E 2 ⋅ e − i ωτ + a 2 = E 1 ⋅ w 1 ′ + E 2 ⋅ w 2 ′ , und mithin U 2 = E ⊤ ⋅ w ′ .
[0065] The amplification factor m and the second overlay parameter a2 should be determined in such a way that the representation limits the angle θ of the maximum attenuation (see Figure 1 ) to a desired angular range.
[0066] 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 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, is now established from equation (x), with the consideration motivated by equations (iii) and (iv'): r e i φ = m e − i ωτ + a 2 1 + a 2 e − i ωτ
[0067] This exploited the fact that the zeros of the polynomial in equation (iii) are only up to a factor c _ ∈ ℂ are defined, resulting in w2 / w1 = w2' / w1'. Since the fraction on the right side of the absolute value is 1, the gain factor is m = r, where r is determined based on the value a1 .0 of the first overlay parameter a1 is given by the absolute value of equation (viii).
[0068] For the second overlay parameter a2 as the second alternative parameter ap2 of the method according to Figure 4 results from equation (xi): a 2 = e − i ωτ − e iφ e i φ − i ωτ − 1 = c o s φ − cos ωτ 1 − cos φ − ωτ
[0069] Using corresponding tabulated values, 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 overlay U1 (e iϕ - iωτ cosθ< = 1, see equation (iv')) and thus also of the second overlay U2 (which initially only represents a conversion of the first overlay U1) can now be established.
[0070] From this, a corresponding adjusted value for the second overlay parameter a2, i.e. an adjusted second overlay parameter a2' or a limited second alternative parameter ap2', can be determined.
[0071] The output signal out can now be formed (if necessary after further signal processing steps 20 and frequency response correction factors 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), but using the adjusted second superposition parameter a2' (instead of, as in equation (i'), using the second superposition parameter a2). The gain factor m in the second front and second rear intermediate signals Z2v, Z2h according to equation (ix) is thereby derived from r = m according to equation (xi) with r according to equation (viii) from the first superposition parameter a1 .
[0072] However, the gain factor m and the adjusted second overlay parameter a2` can also be recalculated 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 superposition based on the adjusted first superposition parameter thus determined 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 U1, is independent of the angle θ of the minimum sensitivity.
[0073] Although the invention has been illustrated and described in detail by the preferred embodiment, the invention is not limited to the disclosed examples and other variations may be derived therefrom by those skilled in the art without departing from the scope of the invention. List of reference symbols
[0074] 1Hearing instrument 2Hearing aid 4Sound signal (from the environment) 5Control unit 6Frontal direction 8Source of interference 10Noise 12Adaptation 20Signal processing steps a1(') (adjusted) first superposition parameter a2(') (adjusted) second superposition parameter ap1, ap2 first or second alternative parameter ap2` limited second alternative parameter c cor correction factor E1, E2 first or second input signal M1, M2 first or second input transducer out output signal m gain factor r quotient (of the magnitudes of the coefficients) U1, U2 first or second superposition w1('), w2(') coefficients Z1v, Z1 first front or first rear intermediate signal Z2v, Z2h second front or second rear intermediate signal ΔθAngle range θAngle (minimum sensitivity) τTime delay ϕRelative phase (of the coefficients)
Claims
1. A 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 formed on the basis of 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 determined by means of a complex-valued first superposition parameter ( a1 ) and based on the first overlay parameter ( a1) is adapted, - whereby a complex value ( a1 .0) of the first overlay 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 via a corresponding modification of the second alternative parameter (ap2), and in the process a modified second alternative parameter is formed, 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).
2. 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 the 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).
3. The 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).
4. 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) in the first superposition (U1), wherein the first alternative parameter (ap1) is formed on the basis of a quotient (r) of the magnitudes 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.
5. The method according to claim 4, wherein based on the first alternative parameter (ap1) and the limited second alternative parameter (ap2`) an adapted first overlay parameter (a1' ), and wherein the superposition for generating the output signal (out) is carried out by the first superposition (U1) on the basis of the adapted first superposition parameter (a1' ) is formed.
6. 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 superposition parameter (a2) and an associated value of a real-valued amplification factor (m), wherein the real-valued amplification factor (m) corresponds to a corresponding amplification of the second input signal (E2) in the formation of the first front or rear signal.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) with amplification of the second input signal (E2) with the said amplification factor (m), the angle (θ) of minimum sensitivity is limited to the predetermined angular range (Δθ), and in this way an adapted second superposition parameter (a2`) is generated, and - wherein the output signal (out) is generated on the basis of the adapted second superposition parameter (a2`) and the amplification factor (m) and on the basis of a superposition of the first input signal (E1) and the second input signal (E2).
7. The method according to claim 5, wherein the second overlay parameter (a2) is used as the second alternative parameter (ap2) and the gain factor (m) is used as the first alternative parameter (ap1).
8. The method according to claim 6 in conjunction with claim 4, wherein the first alternative parameter (ap1) is formed on the basis of the quotient (r) of the magnitudes of the two coefficients (w1, w2) of the first input signal (E1) and 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) to one another, and wherein the adaptation of the second superposition parameter (a2) is carried out on the basis of the first alternative parameter (ap1) and the limited second alternative parameter (ap2'), and thus the adapted second superposition parameter (a2') is formed.
9. Method according to one of claims 6 to 8, wherein the output signal (out) is generated based on the first superposition (U1), for which purpose the value ( a1 .0) of the adjusted first overlay parameter ( a1 ) is determined using the adjusted second overlay parameter (a2`) and the gain factor (m).
10. The method according to claim 8, - wherein a second front intermediate signal (Z2v) and a second rear intermediate signal (Z2h) are formed 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`) 11. 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 by a direction from the second input transducer (M2) to the first input transducer (M1), and - wherein the first front intermediate signal (Z1v) has a relative attenuation in a direction opposite to the frontal direction (6).
12. The method according to claim 11, wherein the first front intermediate signal (Z1v) and the first rear intermediate signal (Z1h) are each generated by means 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).
13. The method according to claim 12, wherein a delay is implemented by means of an all-pass filter in at least one frequency band 14. Method according to one of the preceding claims, wherein in a first adaptation step a first value of the complex first overlay parameter ( a1 ), wherein said first value of the first overlay 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 on the basis of the first alternative parameter (ap1) and the limited second alternative parameter (ap2`) a second value of the first overlay parameter ( a1 ), and wherein said second value of the first overlay parameter ( a1 ) is used for a second adaptation step.
15. Method according to one of the preceding claims, wherein the first overlay parameter ( a1 ) is determined using a least mean squares algorithm and / or a gradient method.
16. Method according to one of the preceding claims, wherein the output signal (out) is additionally superimposed on the first and second input signals (E1, E2) using a correction filter (c cor ) for the frequency response, whereby the correction filter (c cor ) for the frequency response is selected such that the frequency response is flat for the frontal direction (6).
17. 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.