Generation of probe noise in a feedback cancellation system

• US 8,538,052 B2
• Filed: 06/26/2008
• Issued: 09/17/2013
• Est. Priority Date: 07/10/2007
• Status: Active Grant

• Current Assignee: Oticon A/S (William Demant Fonden)
• Original Assignee: Oticon A/S (William Demant Fonden)
• Inventors: Elmedyb, Thomas Bo, Hellgren, Johan
• Primary Examiner(s): Le, Huyen D
• Application Number:
  US12/668,329
• Publication Number:
  US 20100183164A1
• Time in Patent Office: 1,909 Days
• Field of Search: 381/312, 381/317, 381/318, 381/320, 381/321, 381/71.11, 381/83, 381/93
• US Class Current: 381/318
• CPC Class Codes: H04R 25/453 electronically
• CPC Class Codes: H04R 3/02 for preventing acoustic reaction , i.e. acoustic oscillatory feedback specially adapted for hearing aids H04R25/453

Abstract

The invention regards a scheme for generating a probe noise signal to be used in an anti feedback system of an audio system. The audio system comprises e.g. a microphone for capturing an audio signal, an audio signal processor for adaptation of the audio signal and a receiver for generation of an audible signal. According to an embodiment of the invention, a noise signal is injected into the audio signal path between the microphone and the receiver and used for estimating acoustical feedback, the noise signal being generated by the following steps:

• • converting a digitized audio signal to the frequency domain, in order to obtain a series of magnitude and phase values,
  • changing the phase values such that the phase of the resulting signal becomes less correlated, preferably substantially un-correlated, to the original signal,
  • converting the magnitude and phase back to a time domain signal using the changed phase values.

The invention may e.g. be used in a hearing aid, a headset or a pair of headphones.

Claims

1. A method of generating a probe noise signal for use in feedback cancellation in an acoustic system, the method comprising:

capturing a digitized audio signal by storing consecutive values u(n) of the signal;

deriving signal parameters from the captured digitized audio signal for controlling conversion of the captured digitized audio signal from time domain to frequency domain;

determining a size parameter for controlling size of a series of magnitude values to be generated in the frequency domain;

converting the captured digitized audio signal to the frequency domain U(k) by a transformation, whereby the series of magnitude values Mag[U(k)] and phase values Phase[U(k)], are obtained, wherein the number of samples in each transformation is based on a rate of change of the digitized audio signal;

generating a series of artificial phase values Phase′

[U(k)], which are substantially un-correlated to phase values Phase[U(k)] of the captured signal; and

converting the series of corresponding magnitude values Mag[U(k)] and artificial phase values Phase′

[U(k)] by an inverse transformation to a signal in the time domain thereby generating a digitized probe noise signal r(n) which is substantially un-correlated to the original audio signal u(n).

2. A method as claimed in claim 1 further comprising:

storing consecutive values of the digitized probe noise signal r(n).

3. A method as claimed in claim 1, wherein

the artificial phase values of the generated probe noise signal are generated by a random generator.

4. A method as claimed in claim 1, wherein

the artificial phase values of the generated probe noise signal are set to a fixed value or to a number of fixed values, each corresponding to a different frequency range.

5. A method as claimed in claim 1, further comprising:

a windowing-process to reduce border effects when the transformation is applied to a u(n) vector.

6. A method as claimed in claim 1, further comprising:

scaling the magnitude values of the probe noise signal according to the magnitude values Mag[U(k)] of the captured audio signal such that the probe noise signal remains substantially inaudible when added to the captured audio signal and played to the human ear.

7. A method as claimed in claim 6 whereby masking effects are taken into account in order to determine the maximum allowable magnitude values of the probe noise signal such that the probe noise signal remains substantially inaudible when added to the captured audio signal and played to the human ear.

8. A method as claimed in claim 1, further comprising:

scaling the magnitude values of the probe noise signal to remain below the hearing threshold of an ear of a person to whom the signal is presented.

9. A method as claimed in claim 1, wherein

conversion to the frequency domain, the generation of artificial phase values, and the conversion of the magnitude values and artificial phase values back to a time domain signal is performed in overlapping batches, whereby the probe noise signal is generated by adding the generated noise signal from overlapping batches after subjecting each batch to a windowing function.

10. A method as claimed in claim 1, further comprising:

determining a modulation level parameter from the captured signal and using it for generating the probe noise signal.

11. A method as claimed in claim 1 wherein the digitized probe noise signal r(n) is added to the captured audio signal u(n).

12. A method a claimed in claim 1, further comprising:

reducing the size parameter to decrease the number of samples in response to an increase in the rate of change of the digitized audio signal.

13. A method for cancelling feedback in an acoustic system where the acoustic system comprises a microphone, a signal path, a speaker, an adaptive feedback cancellation filter for compensating at least partly a possible feedback signal between the speaker and the microphone, where an adaptive algorithm for generating filter coefficients for the adaptive feedback cancellation filter is used and where a probe noise signal for the adaptive algorithm is generated by:

capturing a digitized audio signal in the time domain from the microphone;

deriving signal parameters from the captured digitized audio signal for controlling conversion of the captured digitized audio signal from time domain to frequency domain;

determining a size parameter for controlling size of a series of magnitude values to be generated in the frequency domain;

transforming the captured digitized audio signal to the frequency domain, whereby a series of magnitude values are obtained, wherein the number of samples in each transformation is based on a rate of change of the digitized audio signal;

generating a series of artificial phase values which are un-correlated with real phase values of the captured signal;

allocating corresponding magnitude values and artificial phase values of the series of values; and

converting the allocated magnitude values and artificial phase values to a time domain signal to obtain a probe noise signal.

14. A method according to claim 13 wherein the probe noise signal is added to the captured digitized audio signal and used as an input for the adaptive algorithm.

15. A hearing aid, comprising:

a probe noise signal generator for use in feedback cancellation in an acoustic system, the probe noise signal generator comprising

an input buffer for storing consecutive values u(n) of a captured, digitized audio signal,

a converting unit for converting the captured, stored audio signal to the frequency domain U(k) by a transformation, whereby a series of magnitude values Mag[U(k)] and phase values Phase[U(k)], are obtained,

a generating unit for generating a series of artificial phase values Phase′

[U(k)], which are un-correlated to phase values Phase[U(k)] of the captured signal, and

an inverse converting unit for converting the series of corresponding magnitude values Mag[U(k)] and artificial phase values Phase′

[U(k)] by an inverse transformation to a signal in the time domain thereby generating a digitized probe noise signal r(n);

an input transducer for converting an input sound to an electric input signal;

an output transducer for converting a processed electric output signal to an output sound;

a forward path defined between the input transducer and the output transducer, the forward path including a signal processing unit defining an input side and an output side of the forward path;

a feedback loop from the output side to the input side comprising a feedback estimation unit for estimating the effect of acoustic feedback from the output transducer to the input transducer, wherein

an input signal to the feedback estimation unit from the output side of the forward path includes the digitized probe noise signal from the probe noise signal generator.

16. A hearing aid according to claim 15, further comprising:

an output buffer for storing consecutive values of the digitized probe noise signal r(n).

17. A hearing aid according to claim 15, wherein

the generating unit comprises a random generator for generating artificial phase values of the generated noise signal.

18. A hearing aid according to claim 15, wherein

the generating unit comprises a fixed value generator for generating artificial phase values of the generated noise signal.

19. A hearing aid according to claim 15, further comprising:

an adding unit for adding the digitized probe noise signal r(n) and the captured, digitized audio signal u(n).

20. A hearing aid according to claim 15, wherein

the feedback estimation unit comprises an adaptive FBC filter comprising a variable filter part for providing a specific transfer function and an update algorithm part for updating the transfer function of the variable filter part,

the update algorithm part receiving first and second update algorithm input signals from the input and output side of the forward path, respectively,

wherein the input signal to the update algorithm part from the output side of the forward path includes the digitized probe noise signal from the probe noise signal generator.

21. A hearing aid according to claim 20 wherein the variable filter part receives an input from the output side of the forward path and delivers an output, which is added to the electric input signal to provide a feedback corrected input signal, which is used as an input to the signal processing unit and to the algorithm part of the adaptive filter.

22. A hearing aid according to claim 20 wherein the input to the update algorithm part from the output side of the forward path is equal to the digitized probe noise signal from the probe noise signal generator.

23. A hearing aid according to claim 20 wherein the input to the variable filter part from the output side includes the digitized probe noise signal from the probe noise signal generator and the output from the signal processing unit.

24. A hearing aid according to claim 20 wherein the input to the update algorithm part from the output side includes the digitized probe noise signal from the probe noise signal generator and the output from the signal processing unit.

25. A hearing aid according to claim 15, wherein

the output from the signal processing unit is an input to the probe noise signal generator.

26. A hearing aid according to claim 15, wherein

the digitized probe noise signal r(n) from the probe noise signal generator is added to the captured, digitized audio signal u(n) and used as an input to the feedback estimation unit.

27. A hearing aid according to claim 26, wherein

the digitized audio signal u(n) is delayed before being added to the digitized probe noise signal r(n) to compensate for a possible delay in the probe noise signal generator.