The present invention is related to a method for determining a maximum gain in a hearing device according to the pre-amble of claim 1 as well as to a hearing device according to the pre-amble of claim 8.
The feedback stability of a hearing device is a crucial variable in the fitting of the hearing device to the users hearing loss and hearing preferences. The feedback stability depends on a couple of factors, e.g. the acoustic path from the receiver back to the microphone, including ear geometry, type of ear shell, vent size, tubing, etc., the mechanical stability of the hearing device housing, especially the mechanical coupling between receiver and microphones, and electromagnetic couplings.
Hence, the fitting software offers (or requires) to test the feedback stability with a feedback test. Different methods are known to perform this task:
A first method is called “Direct Method” and is implemented by increasing the gain in every frequency band until the system becomes unstable. The gain at which just no feedback occurs is then used as the maximum gain in the corresponding frequency band.
A second method is called “Negative-Slope Method” and is described in U.S. Pat. No. 7,010,135 B2. The known technique uses a gain curve (i.e. gain vs. input level) having a negative slope in each band. Because of a high gain at a low input signal, feedback is forced by increasing the input level. As a result thereof, the gain decreases until a stable state is reached. The gain at this stable point is the maximum stable gain, i.e. the feedback threshold.
A third method is called “Open-Loop Identification” and is, for example, described in the publication entitled “Adaptive Filter Theory” by S. Haykin (Prentice Hall, 1996). The loop consisting of a signal processing unit in the hearing device and the feedback path is opened. While applying a probe signal at the output of the signal processing unit of the hearing device, the response at the input of the signal processing unit is measured. By way of correlation (or adaptive filtering) the feedback threshold can be determined. Robustness against environment sound can be achieved through longer averaging times and pseudo-noise techniques.
A fourth method is called “Closed-Loop Identification”: While the hearing device is in normal operation, a probe signal is preferably injected at the output of the hearing device. The identification techniques are the same as for the third method described above. The accuracy is though lower, because of the closed-loop operation. Consequently, the signal level of a necessary probe signal has to be rather high so that it is often perceived as uncomfortably loud.
A fifth method is called “Starkey Destiny” and is disclosed in a paper entitled “Active Feedback Intercept” by S. Banerjee (Starkey White Paper, 2006). A self-learning of a feedback canceller initialisation is mentioned. But in the fitting software that is used to adjust a hearing device, a feedback test with a probe signal has to be done in order to activate the feedback canceller. No other information is disclosed in relation to self-learning features.
The known techniques have at least one of the following drawbacks:
It is therefore one object of the present invention to provide a method for determining a maximum gain in a hearing device, which method does at least not have one of the above-mentioned disadvantages.
This and other objects are accomplished by the measures specified in the characterizing part of claim 1. Additional embodiments of the present invention as well as a hearing device are specified in further claims.
A second problem that can be solved by the invention is the adaptation of this once measured feedback stability over time during the everyday use of the hearing device. The invention allows tracking the long-term changes of the feedback stability.
The present invention is further explained in more detail by referring to drawings illustrating exemplified embodiments of the present invention.
For determining a maximum gain that is adjusted in the gain unit 3, the feedback transfer function F(z) is estimated using an adaptive filter. In fact, the feedback transfer function F(z) in the feedback signal path 15 is estimated to obtain an estimated feedback transfer function F′ (j, k). Thereto, the very well known technique for adaptive feedback canceller is applied using a LMS-(Least Mean Square)-algorithm for minimizing the error of the adaptation.
The LMS-algorithm is implemented in a LMS unit 10, to which the delayed output signal U(j, k) of the limiting unit 6 is fed via a further frequency-to-time domain transfer unit 8. In addition, a difference signal E(j, k), which is fed to the gain unit 3 as well as to an addition unit 13, is also inputted to the LMS unit 10. From the two input signals, coefficients for the estimated feedback path transfer function F′ (j, k) can be calculated. As long as the output signal E(j, k) contains a portion of the feedback signal of the feedback signal path 15, the estimation of the estimated feedback path transfer function F′ (j, k) can be further improved.
The estimated feedback path transfer function F′ (j, k) or Fj[k] mimics the external feedback path 15—i.e. its transfer function F(z)- and can therefore be described by its coefficients, which are called FC coefficients hereinafter. It is pointed out that the adaptive filter can be implemented in the frequency or in the time domain.
The FC coefficients are updated with a fast tracking speed with the adaptive filter algorithm. The movement of the FC coefficients follows each change in the feedback path and also possesses natural fluctuations. In addition, the adaptive filter algorithm is not perfect such that temporarily misadjusted FC coefficients may follow. This is especially true if the loop gain is low, which will be further explained in more detail below.
The shift unit 4 is used to prevent a correlation between the error signal E(j, k) and the signal U(j, k) and basically is a frequency shifter as known, for example, from the paper entitled “Adaptive feedback cancellation with frequency compression for hearing aids” by Harry Alfonso L. Joson et al. (J. Accoust. Soc. Am. 94 (6), December 1993, pp. 3248-3254).
The use of the shift unit 4 further stabilizes the operation of the adaptive filter such that the resulting FC coefficients are less erroneous.
In
The further parameters P might be one or several of the following:
In a first embodiment of the present invention, the preprocessing unit 17 is used to smooth FC coefficient fluctuations, i.e. an averaging of the FC coefficients is performed in the preprocessing unit 17 in order to get rid of fast changing FC coefficients.
The averaged FC coefficients Fj[k], wherein j is the frame number and k is the frequency bin, can be seen as a frequency-domain representation, or the averaged FC coefficients ft[n], wherein t is the time and n is the filter time lag, can be seen as a time-domain representation of the estimated frequency transfer function F′ (j, k) (
In a further embodiment of the present invention, the maximum gains may have to be known in terms of specific frequency bands (e.g. on the Bark scale). The conversion from frequency-domain or time-domain FC coefficients to frequency bands is also done by the conversion unit 18.
A possible processing performed in the conversion unit 18 can be performed using the following formulas:
In time domain:
MSGtdB[b]=−F2B{10 log(|FFT{ft[•]}|2)}
In frequency domain:
MSGjdb[b]=−F2B{10 log(|Fj[k]|2)}
where MSG is an acronym for Maximum Stable Gain and F2B denotes the conversion from frequency bins to Bark bands, wherein a so-called Bark band comprises a collection of adjacent frequency bins. The operation performed on a collection of frequency bins is, for example, an operation to obtain a maximum of the values of specified frequency bins in a Bark band according the following formula, for example:
where lb and hb are the lower and upper border of Bark band b. Other operations, such as mean or median, are also possible.
Although
The control unit 19 steers the preprocessing of the FC coefficients in the preprocessing unit 17. The means of steering comprises a possible freezing of the running preprocessing, an adjustment of the time constants of the preprocessing as well as a (time-dependent) weighting of the FC coefficients prior to the preprocessing.
The preprocessing may be frozen if the (variable) gain in the gain unit 3 is too low, if the difference to the theoretical feedback threshold is too high or if the hearing device is not in operation, which may be detected by an automatic detection unit (not shown in
As has been mentioned above, an averaging of the FC coefficients is performed in the preprocessing unit 17 in one embodiment. In a further embodiment, the preprocessing unit 17 comprises a decision unit that decides, when the preprocessing, for example the averaging of FC coefficients, is to be activated and when frozen, or how the time-constant for the preprocessing is adapted. In another embodiment, a dependency on the prescribed gain is taken into account during the preprocessing step. In still another embodiment, the input level of the acoustic signal is consulted for controlling the preprocessing. In a further embodiments, the variance or the histogram of the FC coefficients are analyzed as well as certain measures derived thereof, as for example percentiles by using dual-slope averaging.
In
In a further embodiment of the present invention, the extraction unit 16 (
It is pointed out that an important aspect and advantage of the present invention is that it can be used during regular operation of the hearing device. Nevertheless, the present invention can also be used during a fitting session, during which an audiologist adjusts a hearing device for a later regular use by the hearing device user. While predefined probe signals are presented to the hearing device user having inserted his hearing device during a fitting session in order to adjust the hearing device, in particular the maximum gain and the threshold level, respectively, the hearing device is continuously adjusted during regular operation using the acoustic signals that are presented during every day usage to the hearing device user. In both applications, there is no need to interrupt regular operation, nor is it necessary to present a certain probe signal.
This embodiment can be used, for example, if feedback threshold estimation is desired before the feedback canceller is activated.
The first use case is as replacement of state-of-the-art feedback threshold estimation methods. The feedback threshold is measured during the fitting session in order to set maximal gains such that the hearing instrument operates in a stable condition. Other optimizations depending on the measured feedback threshold are possible.
Described method can also be used during every-day operation of the hearing instrument. In this use case, the hearing instrument measures the hearing threshold continuously. This continuously measured feedback threshold may be readout by the fitting software in the following fitting session and used as information for the fitter, who can make adjustments based on the continuously measured feedback threshold.
The measured feedback threshold can also be used to adjust parameters of the hearing instrument online and automatically, e.g. reduce the maximum gain if the feedback threshold has worsened.
| Filing Document | Filing Date | Country | Kind | 371c Date |
|---|---|---|---|---|
| PCT/EP08/50701 | 1/22/2008 | WO | 00 | 7/22/2010 |