This application claims priority of German application No. 102005019149.5 filed Apr. 25, 2005, which is incorporated by reference herein in its entirety.
The invention relates to a hearing aid apparatus, particularly to a hearing aid apparatus with compensation of acoustic and electromagnetic feedback signals.
DE 103 13 330 A1 discloses the weighted combination of two microphone signals MS1 and MS2 from microphones M1 and M2, thereby enabling the effect of an acoustic interference signal on the reception of a directional microphone system to be suppressed on a direction-dependent basis.
EP 1 307 072 A2 discloses a hearing aid with three microphones wherein a delay element is connected to one of the microphones. This arrangement is designed to prevent interfering acoustic effects during turn-on, turn-off or switching processes.
Publication EP 1 367 856 A2, however, also discloses an apparatus and a method for electrical feedback reduction in hearing systems. For this purpose an amplifier apparatus is proposed which, in addition to an amplifier device having an acoustic and an inductive input, has two separate compensation paths. In the first compensation path, a first filter device acts to compensate acoustic feedback and in the second compensation path a second filter device acts to compensate inductive feedback. In this way the two very different feedbacks can be individually compensated without using a complex common filter.
The object of the present invention is to improve, in terms of listening convenience, or simplify a hearing aid apparatus comprising a microphone and an electromagnetic receiver as input transducers.
This object is achieved according to the invention by a hearing aid apparatus comprising a microphone and an electromagnetic receiver as input signal transducers, an earpiece as output signal transducer, there being produced acoustic feedback to the microphone and electromagnetic feedback to the electromagnetic receiver or a following electrical component, a signal processing device connected between the input transducers and the output transducer, and an adaptive compensation device, which is connected to the signal processing device and has a single adaptive filter, for compensating the acoustic feedback, the compensation device being used simultaneously for compensating the electromagnetic feedback and a delay element being connected between the electromagnetic receiver and the signal processing device, thereby ensuring that both the acoustic feedback and the electromagnetic feedback is compensated by a single adaptive filter.
There is additionally provided according to the invention a hearing aid apparatus comprising a microphone, an electromagnetic receiver and a signal processing device for processing the signals of the microphone and/or of the electromagnetic receiver as well as a weighting device with which the signal of the microphone and/or of the electromagnetic receiver can be individually weighted prior to processing by the signal processing device so that both signals are audible through the hearing aid apparatus in the appropriate weighting ratio. This means that the hearing aid can be used in purely microphone mode, in purely telephone coil mode and in mixed mode.
For the hearing aid apparatus with the common adaptive filter for compensating the acoustic and electromagnetic feedback, a weighting device is also preferably connected between the input transducers and the signal processing device so that the signals of the input transducers can be individually weighted. This enables compensation to be always achieved in all operating modes (microphone mode, coil mode or mixed mode) using simple means.
In a particular embodiment an adder device can be connected upstream of the signal processing device in order to add the weighted signals. This means that the weighted input signals of the input signal transducers are combined prior to further signal processing.
In another particular embodiment the compensation device can incorporate a timing element whose delay corresponds to the propagation time of the acoustic feedback signal, the delay introduced by the delay element between the electromagnetic receiver and the signal processing device corresponding to the propagation time difference of the acoustic and electromagnetic feedback signal, thereby enabling the length of the adaptive filter to be kept short for digital signal processing.
The present invention will now be explained in greater detail with reference to the accompanying drawings in which:
The present invention relates to a hearing aid apparatus comprising a microphone and an electromagnetic receiver as input signal transducers, an earpiece as output signal transducer, there being produced acoustic feedback to the microphone and electromagnetic feedback to the electromagnetic receiver or a following electrical component, a signal processing device connected between the input transducers and the output transducer, and an adaptive compensation device, which is connected to the signal processing device and has a single adaptive filter, for compensating the acoustic feedback.
In addition to acoustic feedback signals which are overcoupled from the hearing aid output to the microphones and may thus result in feedback whistle, in the case of hearing aids with telephone coils there is additionally the risk, during telephone coil operation, of electrical or more precisely electromagnetic feedback between the hearing aid earpiece and the telephone coil, which may likewise result in feedback whistle. These problems have hitherto been solved mainly by skillful placement and shielding of the components concerned, in particular the hearing aid earpiece.
The examples described below represent preferred embodiments of the present invention.
In the first exemplary embodiment according to
To compensate the acoustic feedback signal, the input signal of the loudspeaker L is fed via a first timing element T1 to an adaptive filter AF1 with the variable transfer function h1. The output signal of the adaptive filter AF1 is subtracted in a first subtractor S1 from the weighted microphone signal. The output signal of the subtractor S1 is used, among other things, for adapting the filter AF1.
Analogously to the acoustic compensation path T1, AF1 there is provided an electromagnetic compensation path T2, AF2 with the second timing element T2 and the adaptive filter AF2 whose transfer function is h2. The output signal of the adaptive filter AF2 is subtracted from the weighted signal of the telephone coil T using a second subtractor S2. Here too, the output signal of the subtractor S2 is used for adapting the second adaptive filter AF2.
The output signals of the subtractors S1 and S2 are added together in an adder A1 and the summation signal is fed to a signal processor SV. The output signal of the signal processor feeds the loudspeaker L.
The electrical feedback between earpiece or loudspeaker L and microphone coil or telephone coil T is therefore compensated in the same way as the acoustic feedback by modeling the transmission path between loudspeaker L and telephone coil T by an adaptive filter AF2 and subtracting the earpiece signal weighted with this filter AF2 from the coil output.
The signals of the input transducers can be weighted with the factors a and b, thereby enabling the relationship between the signals picked up to be influenced. If, for example, a=1, b=0 is set, purely microphone mode is present. If, on the other hand, a=0, b=1 is set, coil mode is present. If the factors a and b are selected otherwise, mixed mode is desired whereby both the signal of the microphone and that of the telephone coil are processed in the signal processor SV and presented via the loudspeaker L.
“Switching” between microphone and coil mode can be effected subject to control via a (telephone) classifier. This detects whether wanted signals—in most cases speech—are present at the microphone M and/or at the telephone coil T and then automatically switches seamlessly if required to the relevant mode provided, e.g. purely microphone mode, purely coil mode or mixed mode.
A second embodiment of the present invention is schematically illustrated in
The output signals from microphone M and telephone coil T are once again weighted by the factors a and b using the weighting units G1 and G2. Between the telephone coil T and the assigned weighting unit G2 there is here additionally provided a timing element T4 in order to allow for the slower acoustic feedback. After weighting in the weighting units G1 and G2 the two signals are added together in an adder A2.
The input signal of the loudspeaker L is fed back via a compensation path which has a timing element T3 and via an adaptive filter AF3. The adaptive filter AF3 possesses the transfer function h. The output signal of the adaptive filter AF3 is subtracted in a subtractor S3 from the output signal of the adder A2. The output signal of the subtractor S3 is fed to the signal processor SV whose output signal feeds the loudspeaker L.
The acoustic and electromagnetic feedback signals are therefore compensated here by the common adaptive filter AF3. For purely microphone mode (a=1, b=0) or purely coil mode (a=0, b=1), the filter AF3 undertakes either adaptation to the feedback R1 with the transfer function h1 or to the feedback R2 with the transfer function h2. Therefore the timing element T3 should either be set to the value of the timing element T1 or to that of the timing element T2 from the example in
Of particular interest, however, is mixed mode (a, b random). Here the adaptive filter AF3 can simultaneously undertake adaptation to both signal paths. For this purpose, the value of the timing element T3 must be set to that of the timing element T1 (T3=T1). In addition, T4=T1−T2 must be set so that the coil signal is delayed by the greater acoustic propagation time (T1>T2).
For digital signal processing and a sampling rate of 20 kHz, T4 must be no more than about 15 samples so that this delay is imperceptible. Alternatively, if the delay T4 is to be avoided, T4=0 and T3=T2 can be set, although then, in the worst case scenario, the length of the adaptive filter AF3 must be increased by up to T1−T2 filter values. This increased length is due to the fact that the two feedback signals with the different propagation times must be compensated by the adaptive filter.
Number | Date | Country | Kind |
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10 2005 019 149 | Apr 2005 | DE | national |
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Number | Date | Country | |
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20060239484 A1 | Oct 2006 | US |