EP1458216A2 - Device and method for adaption of microphones in a hearing aid - Google Patents

Abstract

The system has two microphone input signals, Mic In (2) and Ref Mic (7). The signals are fed to first (1) and second (5) microphones with given transmission functions (s/s-pol-ac1) and (s/s- pol-ac2). The output of the first microphone goes to a compensation filter (3') with a transmission function dependent on the first two transmission functions (s-pol-ac1/s-pol-ac2). The output from this filter goes to a first output Out Mic 1 (4). The output of the second microphone goes to a second output Out Mic 2 (8).

Description

The present invention relates to a mutual method Adaptation of several microphones of one hearing aid. About that the present invention also relates to a corresponding one Device for adapting the microphones.

Hearing impaired people often suffer from reduced communication skills in noise. To improve the signal / noise ratio have been around for some time Directional microphone arrangements used, the benefits for the Hearing impaired is undisputed. Often, either First order systems, i. H. with two microphones, or higher order used. The exclusion from the back received interference signals and the focus on frontal incident sounds enable better communication in Everyday situations.

Directional microphones are sensitive to upsets the transfer functions of the microphones by amount and Phase. The sensitivity to upset increases the order of the directional microphone system and with falling frequency. Such directional microphone systems are used at low frequencies most sensitive.

In this context, document EP 0982971 A2 states that a microphone at low frequencies by one First order high pass can be determined. Accordingly According to FIG. 1, a first microphone 1 can be activated by a Characterize the high pass with the transfer function s / s-pol_ac1. The microphone 1 picks up a first input signal 2. This input signal filtered with the high pass filter of the microphone 1 2 is using a first compensation filter 3 converted into a first microphone output signal 4. The Compensation filter 3 has the transfer function s-pol_ac1 / s-pol_ideal. Both numerator and denominator can be used as Polynomial. The counter polynomial of the compensation filter 3 is chosen so that it corresponds to the denominator polynomial of the acoustic high pass of the microphone 1 corresponds. The denominator polynomial of the compensation filter 3 corresponds to the denominator polynomial the high pass of an ideal microphone. By multiplication of the two transfer functions of the high pass, which characterizes the real microphone 1, and the compensation filter 3 results in a standardization with regard to the ideal microphone and the specific transfer function of the first microphone is compensated.

When looking at hearing aid microphones it has been shown that in a simplified approach especially the one at the bottom Existing acoustic band of the usable frequency band High pass must be examined with regard to upsets. Contamination, aging or changes in the environment have a particularly strong effect on this high pass and thus change Amplitude and frequency response of the microphone in particular critical, middle and lower frequency range. One way to reduce moods caused in this way, consists of the same high pass frequency in all microphone paths to force.

In the same way, the specific high pass with the transfer function s / s-pol_ac2 of the second microphone 5 a second compensation filter 6 with the transfer function s-pol_ac2 / s-pol_ideal compensated so that from the second Microphone input signal 7 a corresponding second microphone output signal 8 arises. Again, the denominator polynomial of the high pass 5 by the counter polynomial of the second Compensation filter 6 eliminated. With these two compensation filters 3 and 6, the fluctuations in the high-pass cutoff frequency from microphone to microphone, particularly when low frequencies lead to phase and amplitude errors would be compensated for by in all microphone paths the same corner frequencies can be set.

A method is described in the further document US Pat. No. 6,272,229 B1 for the relative, adaptive phase adjustment of two microphones roughly outlined. This is a general block diagram specified for an adaptive system. The system includes one Block "acoustical delay compensation", which is in a kind Preprocessing the linear phase difference of the microphones, caused by the signal delay between the microphones is, balances. However, an adaptation regulation is not specified.

Other internal realizations primarily affect the input sensitivity difference the microphones on. Over a time-averaged consideration of the input levels at the microphones can draw conclusions about the input sensitivity of the Microphones are pulled. Assuming that the incident Switching signals are delayed, but with almost that same level can be received by all microphones by comparing the averaged input levels on the microphones adjusted the amplitude of the input sensitivities become.

The object of the present invention is that Simplify compensation of microphone differences in hearing aids.

According to the invention, this object is achieved by a method for mutual adaptation of several microphones of a hearing aid, by measuring a first amplitude of a first output signal from a first of the several microphones in one predetermined frequency range, measuring a second amplitude a second output signal from a second one of the plurality Microphones in the specified frequency range and filtering the first output signal as a function of the first amplitude and the second amplitude, so the difference between the two output signals is reduced.

A device for mutual adaptation of several microphones of a hearing aid, with a first measuring device for measuring a first amplitude a first output signal from a first one of the plurality Microphones in a given frequency range, one second measuring device for measuring a second amplitude a second output signal from a second one of the plurality Microphones in the specified frequency range and a filter device, to the first and second measuring devices is connected to filter the first output signal in Dependence on the first amplitude and the second amplitude, so the difference between the two output signals is reducible.

Compared to the prior art according to FIG 1 can by the invention on a compensation filter in a microphone path, the reference path. One compensation filter each is in every path except the reference path, contain. This means that for example with three microphones to provide a compensation filter in two microphone paths while the third microphone path is the reference path is used.

The predetermined frequency range preferably corresponds to measuring the amplitudes of the two output signals of the microphones a frequency band below 150 Hz. In particular the frequency band is between 40 and 60 Hz or 80 to 120 Hz. This is the area where there are differences in the Corner frequency of the high-pass filter of the microphones is particularly strong to make noticable.

The filtering can be adjusted by a control loop, so that the first and second amplitudes correspond to each other. This makes it possible to change the transfer function over time of the microphones, for example due to dirt or counteract aging effectively.

The compensation filtering can be divided into two partial filterings become. A first partial filtering is done by a Denominator polynomial, which is the high pass frequency of the reference path modeled, realized. A second sub-filter is replaced by a Counter polynomial that is adapted so that the averaged level difference between the microphone paths is minimal. The adaptation takes place through the formation of the amount of the signals instead, which eliminates phase dependency. So that can to a unit like the "acoustical delay" mentioned above compensation "block.

Preferably, the coefficients of the counter polynomial are only dependent on a single parameter. This leads to one little effort for adaptation. Is just the counter polynomial adaptable, this does not lead to identical in principle same microphone signals because of an error between the Characteristic of the reference microphone and that in the denominator polynomial described filter effect can exist. The effect of this good approximation is sufficient to directivity to improve significantly with minimal effort.

An optimal adaptation of the two or more microphones to each other is possible if the denominator polynomial can also be varied. This additional adaptation option also ensures faster adaptation by the control loop.

The amount and / or Phase of the first output signal are modified. In order to the setting of the directional microphone can be improved.

The advantage of an adaptation with the microphone model over an adaptation with a filter that has arbitrary phase functions can reproduce, on the one hand, lies in the simplicity of the Realization. On the other hand, it is fundamentally advantageous to assume a simplified model and the Align compensation specifically to the model.

FIG 1

ein Blockschaltbild zur Kompensation von Verschiebungen von Hochpasseckfrequenzen gemäß dem Stand der Technik;

FIG 2

ein Blockschaltbild zur Kompensation von Verschiebungen von Hochpasseckfrequenzen gemäß der vorliegenden Erfindung;

FIG 3

ein Schaltungsdiagramm einer Kompensationsschaltung gemäß einer ersten Ausführungsform der vorliegenden Erfindung; und

FIG 4

ein Schaltungsdiagramm einer Kompensationsschaltung gemäß einer zweiten Ausführungsform der vorliegenden Erfindung.

The present invention will now be explained in more detail with reference to the accompanying drawings, in which:

FIG. 1

a block diagram for the compensation of shifts of high pass corner frequencies according to the prior art;

FIG 2

a block diagram for the compensation of shifts of high-pass corner frequencies according to the present invention;

FIG 3

a circuit diagram of a compensation circuit according to a first embodiment of the present invention; and

FIG 4

a circuit diagram of a compensation circuit according to a second embodiment of the present invention.

The exemplary embodiments described below represent preferred embodiments of the present invention.

The goal is to have two or more microphones in their electrical and adapt acoustic behavior to one another. each Microphone can be characterized by a characteristic in the low-frequency range acoustic high pass, whose corner frequency is about 50 Hz and an electrical high pass, whose corner frequency is about 100 Hz. Both the acoustic as well as the electrical high passes of the several hearing aid microphones are slightly different and can be adapted to each other in the following way.

According to the block diagram of FIG 2 there is an inventive Compensation for microphone differences in that initially as in the prior art according to FIG 1, the microphone input signal 2 with an acoustic high pass 1 of the first Filtered microphone 1 with the transfer function s / s-pol_ac1 becomes. The subsequent compensation filter 3 'has the transfer function s-pol_ac1 / s-pol_ac2. With this Transfer function becomes the second microphone path, which in FIG 2 is shown below, taken into account. In this The second microphone path is the signal as in the prior art 7 of a reference microphone 5 corresponding to high-pass filtering subjected to the transfer function s / s-pol_ac2. The denominator polynomial of the second acoustic high pass of the second microphone 5 is used to standardize the compensation filter 3 'used in the first microphone path. With this standardization the compensation filter 3 'does not have to be ideal Microphone can be normalized to the first microphone output signal 4 to get. In the second microphone path, this can result in a Compensation filters are dispensed with the second microphone output signal 8 to get.

The compensation filter 3 'has a transfer function with a numerator polynomial s-pol_ac1 and a denominator polynomial s-pol_ac2. With simplified compensation, only the Numerator and not the denominator and the numerator adjusted. The The denominator of the compensation filter 3 'is at a nominal frequency established. In the acoustic case, the nominal frequency is at 50 Hz and in the electrical case at 100 Hz. With this fixed nominal frequency is however only an approximation Compensation possible. This approximate compensation is as mentioned, sufficiently good, for example, for directivity of a directional microphone.

The transformation of such a compensation filter from the analog to the digital range leads to a simple first order IIR filter, which can be represented as follows: P 1 ( X p ) · z + p 0 ( X p ) z + q O

The functions p ₁ and p ₀ and the parameter q ₀ result from the European patent application EP 0982971 A2 mentioned at the beginning. The variable z represents the frequency variable of the microphone input signal. The parameter Xp corresponds to a manipulated variable of the compensation filter. The denominator cannot be varied in this simplified approach.

According to a second embodiment of the present invention, an improved adaptation of the compensation filter results from the fact that the denominator in its transfer function can also be varied as follows by means of a parameter Xq: p 1 ( X p ) · z + p 0 ( X p ) z + q 0 ( X p )

An implementation to adapt the high pass of a microphone according to the first embodiment, in which the denominator of the Transfer function of the compensation filter is fixed shown in FIG 3 as a block diagram. The input unit forms the compensation filter 3 ', which is already related was explained with FIG 2. Input signal is also here the signal 2 of a first microphone, with this representation in contrast to FIG 2 on the reproduction of an acoustic High pass, which represents the microphone, is dispensed with has been. Output signal of the compensation filter 3 ', the low-frequency microphone matching in the present case of acoustic high pass at 50 Hz, that is again Signal 4. This is fed to a multiplication unit 10, in which the signal with a corresponding compensation factor 11 broadband corrected for amplitude can be.

In a subsequent bandpass filter 12 there is a frequency range between 40 and 60 Hz from the output signal of the multiplication unit 10 cut out and a level meter 13 fed. There is the level of the frequency range to be analyzed determined from the signal of the first microphone 2.

In parallel, this becomes a second microphone input signal 8 resulting output signal of an alike not shown second or reference microphone also subjected to bandpass filtering. A band pass 14 also cuts the frequency range between 40 and 60 Hz from the output signal of the microphone and delivers the filtered signal also to a level meter 15.

In a subtraction unit, those from the level meters 13 and 15 measured levels subtracted from each other and the resulting level difference for an update unit for updating the Xp variable. An update of the Xp value should, however, only take place if the microphone signals have a correspondingly high level. To do this, the microphone levels become an input level query block 18 fed, which generates an enable Xp signal when both signal levels exceed a certain threshold. Thereby can be prevented in cases where no acoustic input signals but only microphone noise there is a microphone adaptation. The enable Xp signal is therefore looped through to the Xp update block.

The value Xp possibly updated in block 17 is now to complete the control loop on the compensation filter 3 'delivered. The determination of the Xp value and thus the adaptation of the microphones to one another in the Xp update block 17 can be achieved by a (N) LMS (Normalized Leased Meansquare), whereby an "acoustical delay" block is necessary is.

FIG. 4 shows a circuit diagram of an improved version of a matching circuit. The essential structure corresponds to that of FIG. 3, the function blocks corresponding to one another performing essentially the same functions. Only the compensation filter, which is also denoted by the reference symbol 3 ', has a further signal input with which the denominator polynomial can also be changed via the variable X _q .

In order to be able to change both the numerator and denominator polynomials, the output signal of the input level interrogation block 18, with which it is determined whether the two microphone signals have a sufficiently high level, is forwarded to a switch 19. This switch 19 alternately generates an enable-X _q signal and an enable-X _p signal if it receives an enable-X _p -X _q signal from block 18.

In addition to the X _p update block 17, an Xq update block 20 for changing or updating the X _q value is consequently also provided here. If the switch 19 now emits an enable-X _q signal, the X _q value is changed in accordance with the level difference from the subtractor 16. Otherwise, if the switch 19 emits an enable X _p signal, the X _p value in the X _p update block 17 is changed in accordance with the level difference. If the level difference is less than 0, the X _p or X _q value is changed in one direction, and if the level difference is greater than 0 in the corresponding other direction.

The compensation filter 3 'receives the changed or updated X _p and X _q values as manipulated variables. As in the previous embodiment according to FIG. 3, the different high-pass corner frequencies of the microphones in a narrow frequency range around the corner frequencies mean different averaged output levels of the two microphone signals. This means that the level difference depends directly on the difference in the corner frequencies. To adapt the cutoff frequencies, the difference between the levels is simply formed (power difference).

The total distance of a directional microphone from the microphone input up to the exit is often with other at low frequencies First-order high passes described. In addition to the acoustic High pass the microphone still has an electrical one First-order high-pass filter with a corner frequency of approx. 180 Hz. Another high pass results from a coupling capacitor and input resistance of an IC input stage.

The adaptive methods described above can in principle be applied to all high passes.

Claims (16)

Method for the mutual adaptation of several microphones (1, 5) of a hearing aid,
marked by Measuring (13) a first amplitude of a first output signal from a first of the plurality of microphones (1) in a predetermined frequency range (12), Measuring (15) a second amplitude of a second output signal from a second of the plurality of microphones (5) in the predetermined frequency range (14) and Filtering (3 ') the first output signal as a function of the first amplitude and the second amplitude, so that the difference (16) between the two output signals is reduced. The method of claim 1, wherein the predetermined frequency range (12, 14) one or more frequency bands corresponds below 150 Hz. The method of claim 2, wherein the frequency band (s) Frequency bands between 40 and 60 Hz and / or 80 to 120 Hz. Method according to one of claims 1 to 3, wherein parameters adapted for filtering (3 ') in a control loop be so that the first and second amplitude are each other correspond. Method according to one of claims 1 to 4, wherein the Filter (3 ') by multiplying by a denominator and Counter polynomial takes place. The method of claim 5, wherein only the counter polynomial is varied by regulation. The method of claim 5, wherein the numerator and denominator polynomial can be varied by regulation. Method according to one of claims 1 to 7, wherein by the filtering amount and / or phase of the first output signal is modified. Device for the mutual adaptation of several microphones (1, 5) of a hearing aid,
marked by a first measuring device (13) for measuring a first amplitude of a first output signal from a first of the plurality of microphones (1) in a predetermined frequency range (12), a second measuring device (15) for measuring a second amplitude of a second output signal from a second of the plurality of microphones (5) in the predetermined frequency range (14) and a filter device (3 '), which is connected to the first and second measuring device (13, 15), for filtering the first output signal as a function of the first amplitude and the second amplitude, so that the difference (16) between the two output signals can be reduced is. The apparatus of claim 9, wherein the predetermined frequency range (12, 14) one or more frequency bands corresponds below 150 Hz. The apparatus of claim 10, wherein the frequency band (s) Frequency bands between 40 and 60 Hz and / or 80 to 120 Hz. Device according to one of claims 9 to 11, wherein the Filter device (3 ') can be adapted in this way in a control loop is that first and second amplitude are each other correspond. Device according to one of claims 9 to 12, wherein the Filter device (3 ') by a denominator and numerator polynomial can be modeled. The apparatus of claim 13, wherein only that Counter polynomial is variable. The apparatus of claim 13, wherein the numerator and denominator polynomial are variable. Device according to one of claims 9 to 15, wherein with the filter device amount and / or phase of the first Output signal is modifiable.

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