Method of signal processing for maintaining directional hearing with hearing aids

• US 5,325,436 A
• Filed: 06/30/1993
• Issued: 06/28/1994
• Est. Priority Date: 06/30/1993
• Status: Expired due to Fees

• Current Assignee: House Ear Institute
• Original Assignee: House Ear Institute
• Inventors: Gao, Shawn, Jayaraman, Sriram, Sullivan, Jean, Soli, Sigfrid D.
• Primary Examiner(s): Kuntz, Curtis
• Assistant Examiner(s): TRAN, SINH N
• Application Number:
  US08/085,652
• Time in Patent Office: 363 Days
• Field of Search: 381/68, 381/60, 381/68.6, 381/68.7, 381/24, 381/26, 381/68.1, 128/746, 73/585
• US Class Current: 381/313
• CPC Class Codes: H04R 25/505 using digital signal processing
• CPC Class Codes: H04R 25/70 Adaptation of deaf aid to hearing loss, e.g. initial electronic fitting
• CPC Class Codes: H04R 5/027 Spatial or constructional arrangements of microphones, e.g. in dummy heads
• CPC Class Codes: H04S 2420/01 Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]

Abstract

The insertion effects of hearing aids are determined and compensated to restore the ability to have directional hearing in individuals wearing hearing aids. In one aspect a method involves finding the ratio of the unaided head related transfer function to the aided head related transfer function and then designing a hearing aid filter that is the inverse of that derived insertion effect, thereby restoring the ability to hear interaural differences in aided systems both in level and in time of arrival to improve hearing in the presence of noise. The insertion effects can be derived either through frequency domain analyses, using the above-mentioned transfer function calculations and measurements, or in another aspect through time domain analyses, using optimal filter calculations and measurement obtained using a successive data acquisition system that is subsequently time aligned by recording trigger pulses with the data.

Claims

1. A method for obtaining coefficients of a digital filter for use in compensating effects of a hearing aid, comprising the steps of:

determining an unaided head related transfer function for each ear and for a plurality of azimuth locations of a sound source;

determining an aided head related transfer function for each ear having a hearing aid installed thereat and for the plurality of azimuth locations of the sound source;

finding a minimum phase representation of the unaided head related transfer function;

finding a minimum phase representation of the aided head related transfer function;

calculating the ratio between the unaided minimum phase representation and the aided minimum phase representation to form a target filter response; and

obtaining a plurality of filter coefficients by sampling the target filter response at a plurality of frequency values corresponding to frequency increments in the digital filter.

2. A method according to claim 1, comprising the further steps of detecting a central flat response portion of the unaided head related transfer function, truncating the unaided head related transfer function to retain only the detected flat response portion, and using the truncated unaided head related transfer function in subsequent steps.

3. A method according to claim 1, comprising the further steps of detecting a central, flat response portion of the aided head related transfer function, truncating the aided head related transfer function to retain only the detected flat response portion, and using the truncated aided head related transfer function in subsequent steps.

4. A method according to claim 1, in which the step of finding the minimum phase representation of the unaided head related transfer function includes the steps of characterizing a non-minimum phase component as a bulk time delay, computing a bulk time delay component by least-squares fitting a linear function to unwrapped phase data, determining a slope of a result of the least-squares fitting, converting the slope to a time value, and subtracting the converted time value from the unwrapped phase response.

5. A method according to claim 1, in which the step of finding the minimum phase representation of the aided head related transfer function includes the steps of characterizing a non-minimum phase component as a bulk time delay, computing a bulk time delay component by least-squares fitting a linear function to unwrapped phase data, determining a slope of a result of the least squares fitting, converting the slope to a time value, and subtracting the converted time value from the unwrapped phase response.

6. A method according to claim 1, comprising the further steps of smoothing magnitude and phase components of both aided and unaided head related transfer functions.

7. A method according to claim 6, where in the steps of smoothing are performed using a five-sample moving average with uniform weighing in two smoothing passes.

8. A method recording to claim 1, wherein the step of calculating the ratio includes the step of performing complex division on the aided and unaided head related transfer functions.

9. A method according to claim 1, wherein the step of sampling includes performing least-squares frequency sampling on the target filter response to specify a target amplitude and phase response at a plurality of frequency samples.

10. A method according to claim 9, further including the step of specifying five hundred evenly spaced samples over the bandwidth of the target filter response.

11. A method for selecting filter coefficients in a digital filter for use in compensating loss of directional information to a wearer of a hearing aid, comprising the steps of:

determining an unaided head related transfer function using a binaural manikin for each ear and for a plurality of azimuth locations of a sound source;

determining an aided head related transfer function using a hearing aided binaural manikin for each ear and for the plurality of azimuth locations of the sound source;

finding a minimum phase representation of the unaided head related transfer function;

finding a minimum phase representation of the aided head related transfer function;

finding the ratio of the unaided minimum phase representation to the aided minimum phase representation; and

obtaining a plurality of filter coefficients by sampling the target filter response at a plurality of frequency values corresponding to frequency increments in the digital filter.

12. A method according to claim 11, comprising the further steps of detecting a central flat response portion of the unaided head related transfer function, truncating the unaided head related transfer function to retain only the detected flat response portion, and using the truncated unaided head related transfer function in subsequent steps.

13. A method according to claim 11, comprising the further steps of detecting a central, flat response portion of the aided head related transfer function, truncating the aided head related transfer function to retain only the detected flat response portion, and using the truncated aided head related transfer function in subsequent steps.

14. A method according to claim 11, in which the step of finding the minimum phase representation of the unaided head related transfer function includes the steps of characterizing a non-minimum phase component as a bulk time delay, computing a bulk delay component by least-squares fitting a linear function to unwrapped phase data, determining a slope of a result of the least-squares fitting, converting the slope to a time value, and subtracting the converted time value from the unwrapped phase response.

15. A method according to claim 11, in which the step of finding the minimum phase representation of the aided head related transfer function includes the steps of characterizing a non-minimum phase component as a bulk time delay, computing a bulk time delay component by least-squares fitting a linear function to unwrapped phase data, determining a slope of a result of the least-squares fitting, converting the slope to a time value, and subtracting the converted time value from the unwrapped phase response.

16. A method according to claim 11, comprising the further steps of smoothing magnitude and phase components of both aided and unaided head related transfer functions.

17. A method according to claim 16, where in the steps of smoothing are performed using a five-sample moving average with uniform weighing in two smoothing passes.

18. A method recording to claim 11, wherein the step of finding the ratio includes the step of performing complex division on the aided and unaided head related transfer functions.

19. A method according to claim 11, wherein the step of sampling includes performing least-squares frequency sampling on the target filter response to specify a target amplitude and phase response at a plurality of frequency samples.

20. A method according to claim 19, further including the step of specifying five hundred evenly spaced samples over the bandwidth of the target filter response.

21. A method for obtaining coefficients of a digital filter for use in compensating effects of a hearing aid, comprising the steps of:

producing an audio signal having a predetermined frequency content at a predetermined sound pressure level at a first time;

producing a trigger pulse simultaneously with the audio signal;

detecting the produced signal at an eardrum location in the absence of a hearing aid and recording the detected signal in synchronism with the trigger pulse on a first track of a magnetic tape;

inserting a hearing aid adjacent the eardrum location;

producing the audio signal at a second, later time;

producing the trigger pulse simultaneously with the audio signal the second time;

detecting the produced signal at the eardrum location in the presence of the hearing aid and recording the detected signal in synchronism with the trigger pulse on a second track of a magnetic tape in time alignment with the onset of the recorded signal in the first track by aligning the recorded trigger pulse;

sampling the signals recorded in the first and second tracks; and

calculating digital filter coefficients from the sampled signals using discrete-time Wiener equations.

22. A method according to claim 21, wherein the step of recording the detected signal includes converting the detected signal to a digital signal and controlling the recording in response to the trigger pulse.

23. A method according to claim 21, wherein the step of producing an audio signal comprises producing a white, Gaussian, noise signal.

24. A method according to claim 21, wherein the step of producing an audio signal at a first time and a second later time comprise the steps of producing the audio signal at different azimuths relative to the eardrum location, maintaining the sound pressure level constant, summing all detected signals in the absence of the hearing aid to produce a composite unaided signal, and summing all detected signals in the presence of the hearing aid to produce a composite unaided signal.

25. A method according to claim 24, wherein the step of sampling includes sampling the aided and unaided composite signals to obtain estimates of correlation values for use in the step of calculating.

26. A method according to claim 25, wherein the step of calculating includes computing an auto-correlation matrix R and a cross-correlation matrix P from the sampled signals of the composite aided and unaided signals.

27. A method according to claim 26, wherein the Wiener solution is w=R^-1 P.