HEARING INSTRUMENT AND METHOD FOR DIRECTIONAL SIGNAL PROCESSING OF SIGNALS IN A MICROPHONE ARRAY

Abstract

A method for directional signal processing of signals of a microphone array, including first and second microphones generating first and second input signals from an ambient sound, uses the first input signal to form a reference signal transformed into the frequency domain, thereby generating a frequency-space reference signal. The first and second input signals are transformed into the frequency domain, and a first frequency-space directional signal is formed in the frequency domain using the transformed first and second input signals. A frequency-resolved comparison of the frequency-space reference signal with the first frequency-space directional signal or signal derived therefrom in the frequency domain, is used to generate frequency-dependent first gain factors to generate a time filter in the time domain. The reference signal is filtered by the time filter and an output signal is generated from the reference signal filtered by the time filter.

Claims

1. A method for directional signal processing of signals in a microphone array including at least one first microphone for generating a first input signal from an ambient sound and a second microphone for generating a second input signal from the ambient sound, the method comprising: using the first input signal to form a reference signal; transforming the reference signal into the frequency domain and generating a frequency-space reference signal; transforming each of the first input signal and the second input signal into the frequency domain, and forming a first frequency-space directional signal in the frequency domain using the transformed first input signal and the transformed second input signal; generating frequency-dependent first gain factors by using a frequency-resolved comparison of the frequency-space reference signal with the first frequency-space directional signal or a signal derived from the first frequency-space directional signal in the frequency domain; using the first gain factors to generate a time filter in the time domain; and using the time filter to filter the reference signal, and generating an output signal from the reference signal filtered by using the time filter.

2. The method according to claim 1, which further comprises carrying out the frequency-resolved comparison of the frequency-space reference signal with the first frequency-space directional signal or the signal derived from the first frequency-space directional signal in the frequency domain, by using a spectral division, based on which the frequency-dependent first gain factors are generated.

3. The method according to claim 1, which further comprises generating a second frequency-space directional signal as a signal derived from the first frequency-space directional signal in the frequency domain for the frequency-resolved comparison with the frequency-space reference signal, by applying frequency-dependent second gain factors to the first frequency-space directional signal.

4. The method according to claim 3, which further comprises forming the time filter by using a mapping of the frequency-dependent first gain factors into the time domain.

5. The method according to claim 1, which further comprises: determining frequency-dependent second gain factors for the first frequency-space directional signal; and forming the time filter by using a common mapping of the first gain factors and the second gain factors into the time domain.

6. The method according to claim 3, which further comprises determining the frequency-dependent second gain factors for the first frequency-space directional signal by using at least one of noise suppression or dynamic compression or a hearing impairment to be corrected of a recipient of the output signal.

7. The method according to claim 1, which further comprises forming the reference signal from signal components of the first input signal only.

8. The method according to claim 1, which further comprises forming the reference signal in the time domain from the first input signal and the second input signal as a time directional signal by using directional microphony.

9. The method according to claim 1, which further comprises: providing the microphone array with a third microphone for generating a third input signal from the ambient sound; and transforming the third input signal into the frequency domain, and forming the first frequency-space directional signal in the frequency domain from the transformed third input signal.

10. A method for directional signal processing in a hearing instrument, the method comprising: providing the hearing instrument with a microphone array having at least one first microphone for generating a first input signal from an ambient sound and a second microphone for generating a second input signal from the ambient sound; providing the hearing instrument with a control unit; and generating from the first input signal and the second input signal an output signal of the hearing instrument intended for reproduction, according to claim 1.

11. A hearing instrument, comprising: a microphone array including at least one first microphone for generating a first input signal from an ambient sound and a second microphone for generating a second input signal from the ambient sound; and a control unit configured to carry out the method according to claim 10 by using the first and the second input signals.

Description

BRIEF DESCRIPTION OF THE FIGURES

[0033] FIG. 1 is a block diagram of a hearing instrument with a microphone array and directional signal processing according to the prior art;

[0034] FIG. 2 is a block diagram of an alternative embodiment of the signal processing of the hearing instrument of FIG. 1, according to the prior art;

[0035] FIG. 3 is a block diagram of an embodiment of the directional signal processing of the hearing instrument according to FIG. 1 with reduced latency; and

[0036] FIG. 4 is a block diagram of an embodiment of the directional signal processing with reduced latency, which is an alternative to FIG. 3.

DETAILED DESCRIPTION OF THE INVENTION

[0037] Referring now in detail to the figures of the drawings, in which equivalent parts and dimensions are provided with identical reference signs, and first, particularly, to FIG. 1 thereof, there is seen a schematic block diagram of a hearing instrument 1 which includes a microphone array 2 with a first microphone 4 and a second microphone 6. The signal processing described below of the hearing instrument 1 shown in FIG. 1 is configured according to the prior art. The first microphone 4 is configured to generate a first input signal 8 from an ambient sound 7 that is incident on the microphone array 2. Accordingly, the second microphone 6 is configured to generate a second input signal 10 from the ambient sound 7. It is assumed that a possible pre-amplification and/or digitization of the first or second input signal 8, 10 has already been recorded in the corresponding first or second microphone 4, 6. The first and second input signals 8, 10 are each fed to a first filter bank 12, where they are transformed into the frequency domain so that a transformed first input signal 14 or a transformed second input signal 16 is generated.

[0038] In a directional microphony module 18, a first frequency-space directional signal 20 is formed from the transformed first input signal 14 and the transformed second input signal 16 in the frequency domain. In order to form the first frequency-space directional signal 20, any algorithm suitable for forming a frequency band-dependent directional signal can be used in the directional microphony module 18, i.e. in particular delay-and-sum beamforming, delay-and-subtract beamforming, adaptive differential directional microphony, etc. The first frequency-space directional signal 20 is subjected to noise suppression 22, in which, in particular, a useful signal component and a noise signal component are estimated for each of the individual frequency bands, and a gain factor is determined for each frequency band as a function of the useful signal components or noise signal components, so that frequency bands with a high useful signal component are relatively increased and frequency bands with a high noise signal component are relatively reduced.

[0039] After the noise suppression 22 the resulting signal is fed to an amplification module 24, which in particular can include an AGC for dynamic compression of the frequency band-dependent signal components, and using an appropriate adaptation of frequency band-dependent amplification factors can compensate for an individual hearing impairment of a user of the hearing instrument 1.

[0040] A processed second frequency-space directional signal 26 results from the amplification module 24. This second frequency-space directional signal 26 is fed to a synthesis filter bank 28, which merges the frequency band-dependent signal components of the second frequency-space directional signal 26 and transforms them into the time domain. An output signal 30 resulting from this transformation is converted by a loudspeaker 32 of the hearing instrument 1 into an output sound signal 34. The signal processing described herein, from the first filter bank 12 to the synthesis filter bank 28, is preferably carried out on an appropriately configured signal processor or a processor unit including such a signal processor. Such a processor unit is indicated schematically in FIG. 1 as a control unit 35.

[0041] Through the use of the first filter bank 12 and the synthesis filter bank 28, the output signal 30 inevitably experiences a latency relative to the two input signals 8, 10, and thus relative to the ambient sound 7, which is greater the higher the frequency resolution of the first filter bank 12. Typically, this latency is approximately 4 to 7 ms. Although a significant reduction in the latency can be achieved by reducing the frequency resolution, this will also be at the expense of the beamforming capability as well as the ability to suppress interference signal components in the two input signals 8, 10 by using the noise suppression 22, and, possibly, the ability to adjust the gain in the amplification module 24 individually to suit the user of the hearing instrument 1.

[0042] FIG. 2 shows a block diagram schematically illustrating a variation of the hearing instrument 1 according to FIG. 1, which attempts to solve the described problems of latency that occur as a result of the first filter bank 12 and the synthesis filter bank 28. The signal processing shown in FIG. 2 is also configured according to the prior art in this respect. In the hearing instrument 1 according to FIG. 2, a time directional signal 38 is formed from the first input signal 8 and the second input signal 10 by using a time-domain directional microphony module 36. The time directional signal 38 is transformed into the frequency domain by the first filter bank 12, and a resulting transformed time directional signal 40 is fed to the noise suppression stage 22 and then to the amplification module 24. In this case, frequency band-dependent gain factors gj are determined, which are mapped onto a time filter 44 in the time domain by using a mapping 42, which in particular can include a Fourier transformation. The control unit 35 according to FIG. 1 is not shown in FIG. 2.

[0043] The time filter 44 thus implicitly contains the properties of the frequency band-dependent gain factors gj and their effects on the transformed time directional signal 40, but now in the time domain. Accordingly, the time filter 44 is applied to the (original) time directional signal 38 in the time domain to generate the output signal 30. In particular, the time filter 44 is specified as a minimum-phase filter.

[0044] While the processing of the input signals 8, 10 described on the basis of FIG. 2 can indeed reduce the latency compared to the exemplary embodiment of FIG. 1, in that case, however, due to the necessarily broadband directional microphony in the time domain, which is applied to the input signals 8, 10 in the corresponding module 38, there are no available measures for performing a frequency-selective signal processing (e.g. for noise suppression) during the directional microphony.

[0045] FIG. 3 schematically shows a block diagram of a hearing instrument 1 in which the latency is to be kept as low as possible for an ideally frequency-selective, direction-dependent signal processing. The first input signal 8 is used as a reference signal 46. The effects of frequency-selective directional microphony and noise suppression on the individual frequency bands are determined in a manner yet to be described using the reference signal 46, and thus the time filter 44 to be applied to the reference signal 46 is specified.

[0046] The first input signal 8 as the reference signal 46 and the second input signal 10 are transformed into the frequency domain by the first filter bank 12, in a similar way to the exemplary embodiment shown in FIG. 1, thus generating the transformed first input signal 14 as a frequency-space reference signal 48 or the transformed second input signal 16. Also in the present exemplary embodiment, the directional microphony module 18 is then used to generate the first frequency-space directional signal 20 from the transformed first input signal 14—i.e. the frequency-space reference signal 48—and from the transformed second input signal.

[0047] In order to specify the time filter 44, which is preferably to be specified as a minimum-phase filter, first gain factors g1j are obtained in a frequency band-dependent manner, which are determined by a spectral division 45 of the frequency-space reference signal 48 and of a second frequency-space directional signal 50 derived from the first frequency-space directional signal 20. In particular, the magnitudes of the frequency-space reference signal 48 and the second frequency-space reference signal 50 derived from it, or else quantities derived from the magnitudes, can also be divided frequency band-dependently in order to generate the first gain factors g1j.

[0048] This second frequency-space directional signal 50 is generated by feeding the first frequency-space directional signal 20 to the noise suppression stage 22 and to the amplification module 24, where the first frequency-space directional signal 20 is amplified or attenuated frequency band-dependently by applying second gain factors g2j to the first frequency-space directional signal 20 frequency band-dependently. For example, the second gain factors g2j can be formed in each frequency band from the successive application of the individual factors which were determined respectively in the noise suppression 22 and in the amplification module 24 for the respective frequency band.

[0049] The spectral division 45 determines de facto the manner in which the signal processing applied to the first frequency-space directional signal 20, which takes place in the noise suppression stage 22 and in the amplification module 24, is to be modified or compensated if the input variable is not that first frequency-space directional signal 20, but instead the frequency-space reference signal 48. If the first gain factors g1j resulting from the spectral division 45 were applied to the frequency-space reference signal 48, the resulting signal would correspond in magnitude to the second frequency-space directional signal 50 which results from the application of the noise suppression 22 and the amplification module 24 (or from the second gain factors g2j determined there) to the first frequency-space directional signal 20.

[0050] The first gain factors g1j resulting from the spectral division 45 are now mapped from the frequency domain to the time filter 44 in the time domain using the mapping 42, which is preferably provided by a FIR filter. The time filter 44 is thus the equivalent in the time domain to the just described “modification” or “compensation” of the signal processing of the first frequency-space directional signal 20 which must be applied to the frequency-space reference signal 46. In this respect, the influence of the transformed input signal 16 on the second frequency-space directional signal 50 is also incorporated into the time filter 44 though the spectral division 45. Accordingly, the output signal is generated from an application of the time filter 44 to the frequency-space directional signal 48.

[0051] The time filter 46 in the time domain allows the latency to be kept very low, since latencies which arise e.g. due to the first filter bank 12 do not affect the propagation of the reference signal 46 through the signal flow, but only cause the time filter 44 which is applied to the reference signal 46 to be no longer “current” by the amount of the latency, which can be accepted, however, as a trade-off against the significantly reduced latency of the output signal 30 compared to the exemplary embodiment according to FIG. 1.

[0052] FIG. 4 shows a schematic block diagram of an alternative embodiment of the signal processing described according to FIG. 3, which also contains elements of the exemplary embodiment according to FIG. 2, namely that the time filter 44 is applied to a directional signal in the time domain in a manner to be illustrated below.

[0053] From the first input signal 8 and the second input signal 10, a time directional signal 38 is first generated by using the time-domain directional microphony module 36. This can be effected, for example, by applying a delay of one of the two input signals 8, 10 relative to the other, which can vary over time but always has the same effect on all signal components (and therefore in particular acts in a frequency-independent way). In particular, the time directional signal 38 can also be generated by applying an all-pass filter with a frequency-dependent delay to one of the two input signals 8, 10 in the time-domain directional microphony module 36, so that the time directional signal 38 itself can already exhibit a certain frequency dependency with regard to the directional effect.

[0054] In addition, the first and the second input signals 8, 10 are transformed into the frequency domain by using the first filter bank 12, and the first frequency-space directional signal 20 is generated from the transformed first and second input signals 14, 16 by using the directional microphony module 18 in the frequency domain.

[0055] The time directional signal 36 generated as described above in the present exemplary embodiment serves as the reference signal 46, which is transformed into the frequency domain by using a second filter bank 52, thereby generating the transformed time directional signal 40 as the frequency-space reference signal 48. This and the first frequency-space directional signal 20 are subjected to the spectral division 45 to be compared with each other, thereby determining the first gain factors g1j for the respective frequency bands.

[0056] For the first frequency-space directional signal 20 generated as described above, second gain factors g2j are determined by the noise suppression stage 22 and the amplifier module 24, which would need to be applied appropriately to the first frequency-space directional signal 20 to achieve the noise suppression effect of the noise suppression stage 22 or the amplification effect of the amplification module 24 for the first frequency-space directional signal 20.

[0057] In contrast to the exemplary embodiment according to FIG. 3, however, this noise suppression effect or amplification effect is not achieved directly in the first frequency-space directional signal 20. Rather, the second gain factors g2j corresponding to those effects, which were determined in the noise suppression stage 22 and in the amplification module 24, are now combined with the first gain factors g1j, which were obtained from the spectral division 45 of the frequency-space reference signal 48 and the first frequency-space reference signal 20, and mapped in the time domain onto the time filter 44 by using the mapping 42. The time filter 44 thus determined, which in this case also is preferably configured as a FIR filter, is then applied to the reference signal 46 in the time domain—i.e. to the time directional signal 38—thus generating the output signal 30. Finally, the output signal 30 is converted into the output sound signal 34 by the loudspeaker 32.

[0058] The spectral division 45 in the exemplary embodiment according to FIG. 4 determines to what extent the frequency selective beamforming of the directional microphony module 18 (frequency domain) differs from the broadband beamforming of the time-domain directional microphony module 36, so that the first gain factors g1j generated thereby represent de facto the magnitude in the instantaneous gain by which the transformed time directional signal 40 must be compensated in a frequency band-dependent manner, in order to obtain the intrinsic, direction-sensitive sound characteristic inherent in the first frequency-space directional signal 20. This direction-sensitive sound characteristic, characterized by the first gain factors g1j, is thus mapped together with the noise suppression and amplification effect of the noise suppression stage 22 and the amplification module 24, characterized by the second gain factors g2j, in the time domain onto the time filter 44, so that that sound characteristic and those effects can be achieved by applying the time filter 44 to the equivalent of the transformed time directional signal 40 in the time domain, i.e. precisely to the time directional signal 38.

[0059] By applying the time filter 44 as described, in this exemplary embodiment it is also possible to keep the latency of the output signal 30 very low relative to the two input signals 8, 10 and thus relative to the ambient sound 7, in comparison to the exemplary embodiment according to FIG. 1.

[0060] Although the invention has been illustrated and described in greater detail by using the preferred exemplary embodiments, the invention is not restricted by the examples disclosed and other variations can be derived therefrom by the person skilled in the art without departing from the scope of protection of the invention.

[0061] The following is a summary list of reference numerals and the corresponding structure used in the above description of the invention.

TABLE-US-00001 List of Reference Signs: 1 hearing instrument 2 microphone array 4 first microphone 6 second microphone 7 ambient sound 8 first input signal 10 second input signal 12 first filter bank 14 transformed first input signal 16 transformed second input signal 18 directional microphony module 20 first frequency-space directional signal 22 noise suppression 24 amplification module 26 second frequency-space directional signal 28 synthesis filter bank 30 output signal 32 loudspeaker 34 output acoustic signal 35 control unit 36 time-domain directional microphony module 38 time directional signal 40 transformed time directional signal 42 mapping 44 time filter 45 spectral division 46 reference signal 48 transformed reference signal 50 second frequency-space directional signal 52 second filter bank g1j first gain factors g2j second gain factors gj frequency-band-dependent gain factors