Method for suppressing interference noise in an acoustic system and acoustic system

Abstract

A method for suppressing interference noise in an acoustic system with a microphone that generates an input signal and a loudspeaker that generates an acoustic signal which partially feeds back to the microphone. A first intermediate signal is formed along a primary signal path as a function of the input signal, and an output signal is formed via a frequency distortion. The output signal is coupled into a signal feedback path. A second intermediate signal is formed in the signal feedback path via a decorrelation and used as an input value for an adaptive filter. The adaptive filter generates a compensation signal which compensates the input signal. A third intermediate signal is formed from the input signal and/or compensated input signal, which is used as an input value for the adaptive filter. The output signal is fed to the loudspeaker for reproduction.

Claims

1. A method for suppressing an interference noise in an acoustic system, wherein the acoustic system including at least one microphone and at least one loudspeaker, the at least one microphone generating an input signal and the at least one loudspeaker generating an acoustic signal which partially feeds back to the at least one microphone, the method comprising: forming a first intermediate signal along a primary signal path as a function of the input signal and forming an output signal from the first intermediate signal via a frequency distortion; coupling the output signal out from the primary signal path into a signal feedback path; forming a second intermediate signal in the signal feedback path from the output signal via a decorrelation, inputting the second intermediate signal as an input value for an adaptive filter, generating a compensation signal by the adaptive filter, and feeding the compensation signal to the input signal to form a compensated input signal; forming a third intermediate signal from the input signal and/or from the compensated input signal, and using the third intermediate signal as an input value for the adaptive filter; and feeding the output signal to the at least one loudspeaker for reproduction; wherein the frequency distortion for forming the output signal from the first intermediate signal is a temporally-dependent frequency shift.

2. The method according to claim 1, which comprises time-discretizing the input signal and using a least mean square algorithm as the adaptive filter.

3. The method according to claim 2, which comprises normalizing an increment in the LMS algorithm over the second intermediate signal.

4. The method according to claim 1, wherein the frequency distortion for forming the output signal from the first intermediate signal is a frequency shift.

5. The method according to claim 1, which comprises decorrelating the output signal for forming the second intermediate signal by way of a linear prediction filter.

6. The method according to claim 5, which comprises using time-dependent autocorrelation values of the output signal and/or an error signal based on the input signal for filter coefficients of the linear prediction filter.

7. The method according to claim 6, which comprises adapting the filter coefficients of the linear prediction filter as a function of a decorrelation strength of the frequency distortion.

8. The method according to claim 7, which comprises adapting the filter coefficients of the linear prediction filter as a function of a transfer function of a model of the acoustic system, which includes the at least one microphone and the at least one loudspeaker reproducing the corrected output signal.

9. The method according to claim 5, which comprises adapting filter coefficients of the linear prediction filter as a function of a decorrelation strength of the frequency distortion.

10. The method according to claim 5, which comprises adapting filter coefficients of the linear prediction filter as a function of a transfer function of a model of the acoustic system, which includes the at least one microphone and the at least one loudspeaker reproducing the corrected output signal.

11. An acoustic system, comprising: at least one microphone for generating an input signal; at least one loudspeaker for reproducing an output signal; and a control unit configured to carry out the method according to claim 1 for suppressing an interference noise due to a feedback of the output signal, which is reproduced via the at least one loudspeaker, into the input signal generated by the at least one microphone.

12. The acoustic system according to claim 11, configured as a hearing device.

13. The acoustic system according to claim 11, configured as a hearing aid device.

14. A method for suppressing an interference noise in an acoustic system, wherein the acoustic system including at least one microphone and at least one loudspeaker, the at least one microphone generating an input signal and the at least one loudspeaker generating an acoustic signal which partially feeds back to the at least one microphone, the method comprising: forming a first intermediate signal along a primary signal path as a function of the input signal and forming an output signal from the first intermediate signal via a frequency distortion; coupling the output signal out from the primary signal path into a signal feedback path; forming a second intermediate signal in the signal feedback path from the output signal via a decorrelation, inputting the second intermediate signal as an input value for an adaptive filter, generating a compensation signal by the adaptive filter, and feeding the compensation signal to the input signal to form a compensated input signal; forming a third intermediate signal from the input signal and/or from the compensated input signal, and using the third intermediate signal as an input value for the adaptive filter; and feeding the output signal to the at least one loudspeaker for reproduction; wherein the frequency distortion for forming the output signal from the first intermediate signal is a temporally-dependent frequency shift.

15. The method according to claim 14, which comprises time-discretizing the input signal and using a least mean square algorithm as the adaptive filter.

16. The method according to claim 15, which comprises normalizing an increment in the LMS algorithm over the second intermediate signal.

17. The method according to claim 16, which comprises decorrelating the output signal for forming the second intermediate signal by way of a linear prediction filter.

18. The method according to claim 17, which comprises using time-dependent autocorrelation values of the output signal and/or an error signal based on the input signal for filter coefficients of the linear prediction filter.

19. The method according to claim 18, which comprises adapting the filter coefficients of the linear prediction filter as a function of a decorrelation strength of the frequency distortion.

20. The method according to claim 19, which comprises adapting the filter coefficients of the linear prediction filter as a function of a transfer function of a model of the acoustic system, which includes the at least one microphone and the at least one loudspeaker reproducing the corrected output signal.

21. The method according to claim 17, which comprises adapting filter coefficients of the linear prediction filter as a function of a decorrelation strength of the frequency distortion.

22. The method according to claim 17, which comprises adapting filter coefficients of the linear prediction filter as a function of a transfer function of a model of the acoustic system, which includes the at least one microphone and the at least one loudspeaker reproducing the corrected output signal.

23. A method for suppressing an interference noise in an acoustic system, wherein the acoustic system including at least one microphone and at least one loudspeaker, the at least one microphone generating an input signal and the at least one loudspeaker generating an acoustic signal which partially feeds back to the at least one microphone, the method comprising: forming a first intermediate signal along a primary signal path as a function of the input signal and forming an output signal from the first intermediate signal via a frequency distortion; coupling the output signal out from the primary signal path into a signal feedback path; forming a second intermediate signal in the signal feedback path from the output signal via a decorrelation, inputting the second intermediate signal as an input value for an adaptive filter, generating a compensation signal by the adaptive filter, and feeding the compensation signal to the input signal to form a compensated input signal; forming a third intermediate signal from the input signal and/or from the compensated input signal, and using the third intermediate signal as an input value for the adaptive filter; and feeding the output signal to the at least one loudspeaker for reproduction; and decorrelating the output signal for forming the second intermediate signal by way of a linear prediction filter; and adapting the filter coefficients of the linear prediction filter as a function of a decorrelation strength of the frequency distortion; wherein the frequency distortion for forming the output signal from the first intermediate signal is a temporally-dependent frequency shift.

24. The method according to claim 23, which comprises time-discretizing the input signal and using a least mean square algorithm as the adaptive filter.

25. The method according to claim 24, which comprises normalizing an increment in the LMS algorithm over the second intermediate signal.

26. The method according to claim 23, wherein the frequency distortion for forming the output signal from the first intermediate signal is a frequency shift.

27. The method according to claim 23, which comprises using time-dependent autocorrelation values of the output signal and/or an error signal based on the input signal for filter coefficients of the linear prediction filter.

28. The method according to claim 23, which comprises adapting the filter coefficients of the linear prediction filter as a function of a transfer function of a model of the acoustic system, which includes the at least one microphone and the at least one loudspeaker reproducing the corrected output signal.

29. The method according to claim 23, which comprises adapting filter coefficients of the linear prediction filter as a function of a decorrelation strength of the frequency distortion.

30. The method according to claim 23, which comprises adapting filter coefficients of the linear prediction filter as a function of a transfer function of a model of the acoustic system, which includes the at least one microphone and the at least one loudspeaker reproducing the corrected output signal.

Description

BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWING

(1) FIG. 1 shows a block diagram of the sequence of a method for suppressing an interference noise in an acoustic system; and

(2) FIG. 2 shows a block diagram of an additional possible embodiment of the method according to FIG. 1.

DETAILED DESCRIPTION OF THE INVENTION

(3) Referring now to the figures of the drawing in detail and first, particularly, to FIG. 1 thereof, there is shown a schematic block diagram of the sequence of a method 1 for suppressing an interference noise g in an acoustic system 2. The acoustic system 2, which is provided here by a hearing device 3, for example, a hearing aid device, comprises a microphone 4 and a loudspeaker 6. The microphone signal m picked up by the microphone 4 is fed to a signal processing unit 10 in a primary signal path 8, where it is amplified, among other things. At the end of the primary signal path 8, an output signal xs is output to the loudspeaker 6, which generates an acoustic signal p from the output signal xs. A portion of the acoustic signal p generated by the loudspeaker 6 is again picked up by the microphone 4 as feedback fb, and is thus introduced into the microphone signal m. Signal components of the acoustic signal p in the microphone signal m are again fed to the signal processing unit 10 via the feedback fb and further amplified there. Via the repeated amplification, reproduction, and pickup in a closed process, interference noises g in the form of nearly single-frequency whistling tones result. To suppress the interference noises g, the signal feedback path 16 is provided.

(4) For the signal feedback path 16, the output signal xs is decoupled from the primary signal path 8 and fed to a decorrelator 18. In this case, the decorrelator 18 is formed by a linear prediction filter 20.

(5) In the primary signal path 8, the signal processing unit 10 outputs a first intermediate signal x which is converted via a frequency distortion 22 into the output signal xs. The frequency distortion 22, which is achieved in the present case via a frequency shift 23, results in the linear prediction filter 20 not decorrelating the signal components corresponding to the interference noises g, but rather only signal components of a target signal. A second intermediate signal xw is output by the linear prediction filter 20 as an input value to an adaptive filter 24. The adaptive filter 24 generates a compensation signal c from the output signal xs, which is subtracted from the microphone signal m for compensating for the interference noises g. The signal feedback path 16 is thereby closed.

(6) For generating the compensation signal c, an additional intermediate signal ew is fed to the adaptive filter 24 as an input signal. This third intermediate signal ew is formed from the error signal e which results from the microphone signal m compensated by the compensation signal c. The error signal e is now likewise decorrelated via a linear prediction filter 26, and the decorrelated error signal ew is fed to the adaptive filter 24 as a second input value. The coefficients h are now calculated from the decorrelated error signal ew and the second intermediate signal xw in a filter block 28 of the adaptive filter 24, from which a signal block 30 of the adaptive filter generates the compensation signal c in conjunction with the output signal xs.

(7) It is thus ensured via the frequency shift 23 that the linear prediction filter 20 does not decorrelate any signal components belonging to the interference noises g, whereby the adaptive filter 24 would no longer compensate for them with the compensation signal c. The length of the stationary time window T of the linear prediction filters 20, 26, and thus their adaptation speed, is controlled as a function of the frequency shift 23. A control unit 32 in the hearing device 3 carries out all specified method steps.

(8) In FIG. 2, a slight modification of the method 1 depicted in FIG. 1 is shown in a block diagram. Here, in the acoustic system 2, i.e., in particular in a hearing device 3, for example, in a hearing aid device, the decorrelated error signal ew, which is fed to the adaptive filter as an input value, is formed from an input signal mw decorrelated in the linear prediction filter 26 and a decorrelated compensation signal cw. The decorrelated compensation signal cw is formed in the filter block 28 of the adaptive filter from the error signal ew decorrelated in the linear prediction filter 26 and the second intermediate signal xw, which is provided by the output signal xs decorrelated in the linear prediction filter 20. The length of the stationary time window T of the linear prediction filters 20, 26, and thus their adaptation speed, is determined via an adaptation controller 34 into which the degree df of the frequency shift 23, the gain n of the signal processing unit 10 in individual sub-bands, and a transfer function of the acoustic system 2 which is not depicted in greater detail, are introduced and are used for determining the time window T. Likewise, in this case, a model of the acoustic feedback path fb determined via the filter coefficients h may also be used, so that the adaptation speed of the decorrelation in the linear prediction filters 20, 26 is also determined as a function of the feedback estimated via this model. The use of such an adaptation controller 34 is not limited to the form of the signal feedback path 16 depicted in FIG. 2, but may be used in principle in various embodiment variants, in particular in the exemplary embodiment shown in FIG. 1.

(9) Although the present invention was illustrated and described in detail via the preferred exemplary embodiment, the present invention is not limited by this exemplary embodiment. Other variations may be derived from it by those skilled in the art without departing from the scope of protection of the present invention.