Systems and methods for interference-suppression with directional sensing patterns
원문보기
IPC분류정보
국가/구분
United States(US) Patent
등록
국제특허분류(IPC7판)
H04R-025/00
H04R-003/00
H04R-011/04
H04R-011/00
출원번호
US-0409969
(2003-04-09)
발명자
/ 주소
Feng,Albert S.
Lockwood,Michael E.
Jones,Douglas L.
Bilger, legal representative,Carolyn J.
Lansing,Charissa R.
O'Brien,William D.
Wheeler,Bruce C.
Bilger, deceased,Robert C.
출원인 / 주소
Board of Trustees for the University of Illinois
대리인 / 주소
Krieg DeVault LLP
인용정보
피인용 횟수 :
13인용 특허 :
100
초록▼
System (10) is disclosed including an acoustic sensor array (20) coupled to processor (42). System ( 10) processes inputs from array (20) to extract a desired acoustic signal through the suppression of interfering signals. The extraction/suppression is performed by modifying the array (20) inputs in
System (10) is disclosed including an acoustic sensor array (20) coupled to processor (42). System ( 10) processes inputs from array (20) to extract a desired acoustic signal through the suppression of interfering signals. The extraction/suppression is performed by modifying the array (20) inputs in the frequency domain with weights selected to minimize variance of the resulting output signal while maintaining unity gain of signals received in the direction of the desired acoustic signal. System ( 10) may be utilized in hearing, cochlear implants, speech recognition, voice input devices, surveillance devices, hands-free telephony devices, remote telepresence or teleconferencing, wireless acoustic sensor arrays, and other applications.
대표청구항▼
What is claimed is: 1. An apparatus, comprising: a hearing aid input arrangement including a number of sensors each responsive to detected sound to provide a corresponding number of sensor signals, the sensors each having a directional response pattern with a maximum response direction and a minimu
What is claimed is: 1. An apparatus, comprising: a hearing aid input arrangement including a number of sensors each responsive to detected sound to provide a corresponding number of sensor signals, the sensors each having a directional response pattern with a maximum response direction and a minimum response direction that differ in sound response level by at least 3 decibels at a selected frequency, a first axis coincident with the maximum response direction of a first one of the sensors being positioned to intersect a second axis coincident with the maximum response direction of a second one of the sensors at an angle in a range of about 10 degrees through about 180 degrees; and a hearing aid processor operable to execute an adaptive beamformer routine with the sensor signals and generate an output signal representative of sound emanating from a selected source, wherein the routine is executable to adjust a correlation factor to control beamwidth as a function of frequency to reduce variance of the output signal and provide the output signal with a predefined gain. 2. The apparatus of claim 1, wherein the sensors are a pair of matched microphones and the directional response pattern is of a cardioid, hypercardioid, supercardioid, or figure-8 type. 3. The apparatus of claim 1, wherein the angle is about 90 degrees. 4. The apparatus of claim 1, wherein the angle is about 180 degrees with the maximum response direction of the first one of the sensors being generally opposite the maximum response direction of the second one of the sensors. 5. The apparatus of claim 1, further comprising a reference axis, the routine being operable to determine the selected source relative to the reference axis. 6. The apparatus of claim 5, wherein the reference axis generally bisects the angle. 7. The apparatus of claim 1, further comprising one or more analog-to-digital converters and at least one digital-to-analog converter, the routine being operable to transform input data from a time domain form to a frequency domain form, and is further operable to adaptively change a number of signal weights for each of a number of different frequency components to provide the output signal. 8. The method of claim 1, wherein the first one of the sensors is spaced apart from the second one of the sensors by a separation distance of less than 0.2 centimeter. 9. A method, comprising: providing a number of sensors each responsive to detected sound to provide a corresponding number of sensor signals, the sensors each having a directional response pattern with a maximum response direction and a minimum response direction that differ in sound response level by at least 3 dB at a selected frequency, a first axis coincident with the maximum response direction of a first one of the sensors being positioned to intersect a second axis coincident with the maximum response direction of a second one of the sensors at an angle in a range of about 10 degrees through about 180 degrees; processing signals from each of the sensors with a hearing aid as a function of a number of signal weights adaptively recalculated from time-to-time; determining a level of interference and adjusting beamwidth in accordance with the level of interference; and providing an output of the hearing aid based on said processing, the output being representative of sound emanating from a selected source. 10. The method of claim 9, wherein the angle is approximately 180 degrees. 11. The method of claim 9, wherein the maximum response direction of the first one of the sensors and the maximum response direction of the second one of the sensors are approximately opposite one another. 12. The method of claim 9, wherein the angle is between about 20 degrees and about 160 degrees. 13. The method of claim 9, wherein said processing includes determining the selected sound source position relative to a reference axis that approximately bisects the angle. 14. The method of claim 9, wherein said processing is further performed as a function of a number of different frequencies. 15. The method of claim 14, which includes varying beamwidth as a function of the frequencies. 16. The method of claim 9, which includes adaptively changing a correlation length. 17. The method of claim 9, wherein the number of sensors is two or more, and the first one of the sensors is approximately collocated with the second one of the sensors to reduce response time difference therebetween. 18. The method of claim 9, wherein the first one of the sensors is spaced apart from the second one of the sensors by a separation distance of less than 0.2 centimeter. 19. An apparatus, comprising: a sound input arrangement including a number of microphones oriented in relation to a reference axis and operable to provide a number of microphone signals representative of sound, the microphones each having a directional sound response pattern with a maximum response direction, the microphones being positioned in a predefined positional relationship relative to one another with a separation distance of less than 0.2 centimeter to reduce a difference in time of response between the microphones for sound emanating from a source closer to one of the microphones than another of the microphones; and a processor responsive to the microphones to generate an output signal as a function of a number of signal weights for each of a number of different frequencies, the signal weights being adaptively recalculated with the processor from time-to-time. 20. The apparatus of claim 19, wherein the microphones include a pair of matched cardioid, hypercardioid, supercardioid, or figure-8 microphones. 21. The apparatus of claim 19, wherein an angle between the maximum response direction of a first one of the microphones relative to a second one of the microphones is in a range of about 10 degrees through about 180 degrees and the processor is further operable to generate the output signal relative to the reference axis and the reference axis approximately bisects the angle. 22. The apparatus of claim 19, wherein the processor includes means for adjusting a factor to control beamwidth as a function of frequency to reduce variance of the output signal and to provide the output signal with a predefined gain. 23. An apparatus, comprising: a sound input arrangement including a number of microphones operable to provide a number of microphone signals representative of sound, at least a first one of the microphones having a directional sound response pattern with a maximum response direction and a minimum response direction that differ in sound response level by at least 3 dB at a selected frequency and at least a second one of the microphones having an omnidirectional response pattern, the first one of the microphones and the second one of the microphones being positioned relative to one another with a separation distance of less than two centimeters to reduce a difference in time of response between the microphones for sound emanating from a source closer to one of the microphones than another of the microphones; and a processor responsive to the microphones to generate an output signal as a function of a number of signal weights for each of a number of different frequencies, the signal weights being adaptively recalculated with the processor from time-to-time, the processor including means for adjusting a factor to control beamwidth as a function of frequency to reduce variance of the output signal and to provide the output signal with a predefined gain. 24. The apparatus of claim 23, further comprising an output device responsive to the output signal to generate an output representative of sound emanating from a selected source. 25. The apparatus of claim 23, wherein the separation distance is less than about 0.2 centimeter. 26. A method, comprising: providing a number of sensors each responsive to detected sound in a broadband frequency range of at least ⅓ of an octave to provide a corresponding number of sensor signals, one or more of the sensors having a directional response pattern with a maximum response direction and a minimum response direction that differ in sound response level by at least 3 dB at a selected frequency, and at least one other of the sensors having an omnidirectional response pattern; processing signals from each of the sensors with a beamformer routine, said processing including adaptively recalculating several signal weights from time-to-time for each of a number of different frequencies which includes adaptively changing a correlation length to control beamwidth as a function of a number of different frequencies; and providing an output based on said processing, the output being representative of sound emanating from a selected source. 27. The method of claim 26, which includes varying beamwidth as a function of the frequencies. 28. The method of claim 26, which includes utilizing the output in at least one of hands-free telephony equipment, a hearing aide, remote telepresence equipment, an audio surveillance device, speech recognition, a cochlear implant, or a wireless acoustic sensor array. 29. The method of claim 26, wherein a first one of the sensors is spaced apart from a second one of the sensors by a separation distance of less than 0.2 centimeter. 30. An apparatus, comprising: a sound input arrangement including a number of microphones oriented in relation to a reference axis and operable to provide a number of microphone signals representative of sound, the microphones each having a directional sound response pattern with a maximum response direction, the microphones being positioned in a predefined positional relationship relative to one another with a separation distance of less than two centimeters to reduce a difference in time of response between the microphones for sound emanating from a source closer to one of the microphones than another of the microphones; and a processor responsive to the microphones to generate an output signal as a function of a number of signal weights for each of a number of different frequencies, the signal weights being adaptively recalculated with the processor from time-to-time, wherein the processor includes means for adjusting a factor to control beamwidth as a function of frequency to reduce variance of the output signal and to provide the output signal with a predefined gain. 31. The apparatus of claim 30, wherein the microphones include a pair of matched cardioid, hypercardioid, supercardioid, or figure-8 microphones. 32. The apparatus of claim 30, wherein an angle between the maximum response direction of a first one of the microphones relative to a second one of the microphones is in a range of about 10 degrees through about 180 degrees and the processor is further operable to generate the output signal relative to the reference axis and the reference axis approximately bisects the angle. 33. The apparatus of claim 30, further comprising an output device responsive to the output signal to generate an output representative of sound emanating from a selected source.
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