Auditory Front End Customization

• US 20080085023A1
• Filed: 09/25/2006
• Published: 04/10/2008
• Est. Priority Date: 09/25/2006
• Status: Active Grant

• Current Assignee: Advanced Bionics AG (Sonova Holding AG)
• Original Assignee: Advanced Bionics LLC (Sonova Holding AG)
• Inventors: Kulkarni, Abhijit, Mishra, Lakshimi Narayan, Faltys, Michael A.
• Granted Patent: US 7,864,968 B2
• Time in Patent Office: Days
• US Class Current: 381/320
• CPC Class Codes: A61N 1/36038 Cochlear stimulation
• CPC Class Codes: H04R 1/406 microphones
• CPC Class Codes: H04R 2225/021 Behind the ear [BTE] hearing aids
• CPC Class Codes: H04R 2225/67 Implantable hearing aids or parts thereof not covered by H04R25/606
• CPC Class Codes: H04R 2430/20 Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic H04R2203/12 takes precedence
• CPC Class Codes: H04R 25/606 acting directly on the eardrum, the ossicles or the skull, e.g. mastoid, tooth, maxillary or mandibular bone, or mechanically stimulating the cochlea, e.g. at the oval window
• CPC Class Codes: H04R 25/70 Adaptation of deaf aid to hearing loss, e.g. initial electronic fitting

Abstract

A method and system for implementing an acoustical front end customization are disclosed. The customization is implemented to optimize the sound level for each individual cochlear implant user. A known audio signal is generated using a sound source and captured by a microphone system. The captured sound signal is sampled at one or more locations along the signal processing pathway, and a spectrum is determined for the sampled signal and the known signal. A ratio of the two spectrums is related to the undesired transformation of the sampled signal, and a digital filter is designed based on the ratio to filter out the undesired transformation.

Claims

1. A method of customizing a cochlear implant front end, the method comprising:

receiving a known audio signal;

sampling the received audio signal at one or more locations along one or more signal paths to generate a sampled audio signal;

comparing the sampled audio signal against the known audio signal to detect an undesired transformation of the sampled audio signal;

based on the comparison, generating a digital filter operative to minimize the undesired transformation of the sampled audio signal; and

providing the generated filter in the one or more signal paths.

2. The method of claim 1, further comprising:

generating the known audio signal through an external sound source.

3. The method of claim 1, further comprising processing the received audio signal, wherein processing the received signal comprises:

converting the received audio signal into an electrical signal;

converting the electrical signal into a digital signal; and

adjusting a gain of the digital signal.

4. The method of claim 1, wherein comparing the sampled audio signal against the known audio signal further comprises:

determining a spectrum of the sampled audio signal;

determining a spectrum of the known audio signal; and

generating a ratio of the sampled audio signal spectrum over the known audio signal spectrum.

5. The method of claim 4, wherein the digital filter is generated based on the determined ratio.

6. The method of claim 1, further comprising sampling the received signal at two or more locations along the signal path.

7. The method of claim 1, further comprising processing the captured audio signal along two or more signal paths in parallel.

8. The method of claim 7, wherein the digital filter is generated to optimally match a first response of a first microphone with a second response of a second microphone.

9. The method of claim 8, further comprising:

combining the matched first and second responses of the first and second microphones to generate a beamforming signal; and

compensating for a spectral modulation of the beamforming signal.

10. A system for customizing a cochlear implant front end, the system comprising:

a signal generator operable to generate a known audio signal;

a microphone system operable to capture the generated audio signal;

a signal processing system operable to process the captured audio signal along one or more signal paths;

a signal sampler operable to sample the processed audio signal at one or more locations along the signal path;

a signal comparator operable to compare the sampled audio signal against the generated known audio signal to detect an undesired transformation of the sampled audio signal;

a signal filter operable to filter the sampled audio signal based on the comparison to minimize the undesired transformation of the sampled audio signal; and

a signal output generator operable to generate an output signal.

11. The system of claim 10, wherein the signal generator comprises an external sound source.

12. The system of claim 10, wherein the signal processing system further comprises:

an auditory front end unit operable to convert the received audio signal into an electrical signal;

an analog to digital converter operable to convert the electrical signal into a digital signal; and

an automatic gain control operable to adjust a gain of the digital signal.

13. The system of claim 12, wherein the signal sampler is further operable to sample the processed audio signal before adjusting the gain of the digital signal.

14. The system of claim 12, wherein the signal sampler is further operable to sample the processed audio signal after adjusting the gain of the digital signal.

15. The system of claim 10, wherein the signal comparator is further operable to:

determine a spectrum of the sampled audio signal;

determine a spectrum of the known audio signal; and

generate a ratio of the sampled audio signal spectrum over the known audio signal spectrum.

16. The system of claim 15, wherein the signal filter is operable to filter the sampled audio signal based on the determined ratio.

17. The system of claim 10, wherein the signal sampler is further operable to sample the processed signal at two or more locations along the one or more signal paths.

18. The system of claim 10, wherein the microphone system comprises two or more microphones, and the signal processing system is further operable to process the received audio signal along two or more signal paths in parallel.

19. The system of claim 18, wherein the signal filter is operable to optimally match a first response of a first microphone with a second response of a second microphone.

20. The system of claim 19, further comprising:

a beamforming module operable to combined the matched first and second responses of the first and second microphones to generate a beamforming signal; and

an optimization module configured compensate for a spectral modulation of the beamforming signal.

21. A machine readable medium storing instructions operative to cause a data processing apparatus to perform operations comprising:

receiving a known audio signal;

sampling the received audio signal at one or more locations along one or more signal paths to generate a sampled audio signal;

comparing a spectrum of the sampled audio signal against a spectrum of the known audio signal to detect an undesired transformation of the sampled audio signal;

based on the comparison, generating a digital filter to minimize the undesired transformation of the sampled audio signal; and

providing the generated digital filter in the one or more signal paths.

22. The machine readable medium of claim 21, wherein the instructions are further operative to cause a data processing apparatus to generate the digital filter to optimally match a first response of a first microphone with a second response of a second microphone.

23. The machine readable medium of claim 22, wherein the instructions are further operative to cause a data processing apparatus to perform operations comprising:

combining the matched first and second responses of the first and second microphones to generate a beamforming signal; and

compensating for a spectral modulation of the beamforming signal.