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
UP-0484838
(2006-07-11)
등록번호
US-7577266
(2009-08-31)
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, Robert C.
출원인 / 주소
The Board of Trustees of the University of Illinois
대리인 / 주소
Krieg DeVault LLP
인용정보
피인용 횟수 :
3인용 특허 :
102
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 sound processing 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 direc
What is claimed is: 1. An sound processing 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. 2. The sound processing 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 sound processing apparatus of claim 1, wherein the angle is about 90 degrees. 4. The sound processing 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 sound processing 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 sound processing apparatus of claim 5, wherein the reference axis generally bisects the angle. 7. The sound processing 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 sound processing apparatus of claim 1, wherein the routine is executable to adjust a correlation factor to control beamwidth as a function of frequency. 9. A sound processing 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; 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 sound processing method of claim 9, wherein the angle is approximately 180 degrees. 11. The sound processing method of claim 10, 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 sound processing method of claim 9, wherein the angle is between about 20 degrees and about 160 degrees. 13. The sound processing 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 sound processing method of claim 9, wherein said processing is further performed as a function of a number of different frequencies. 15. The sound processing method of claim 9, which includes varying beamwidth as a function of the frequencies. 16. The sound processing method of claim 9, which includes adaptively changing a correlation length. 17. The sound processing 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. An sound processing 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 define an adaptive beamformer 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 based on an amplitude difference between the microphone signals for each of the different frequencies. 19. The sound processing apparatus of claim 18, wherein the processor includes means for adjusting beamwidth in accordance with sound interference level. 20. A sound processing 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 the sensor signals as a function of a number of signal weights adaptively recalculated from time-to-time to define an adaptive beamformer; providing an output based on said processing, the output being representative of sound emanating from a selected source; determining a level of interference; and adjusting beamwidth of the beamformer in accordance with the level of interference. 21. The sound processing method of claim 20, wherein the angle is approximately 180 degrees. 22. The sound processing method of claim 20, wherein the angle is between about 20 degrees and about 160 degrees. 23. The sound processing method of claim 20, wherein said processing includes determining the selected sound source position relative to a reference axis that approximately bisects the angle. 24. The sound processing method of claim 20, which includes varying beamwidth as a function of the frequencies. 25. The sound processing method of claim 20, which includes adaptively changing a correlation length. 26. The sound processing method of claim 20, wherein the first one of the sensors is approximately collocated with the second one of the sensors to reduce response time difference therebetween for sound emanating from a source closer to one of the sensors than another of the sensors, and the signal weights are determined in accordance with an amplitude difference between the sensor signals for each of a number of different frequencies. 27. The sound processing apparatus of claim 9, which includes: positioning the sensors 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 sensors for sound emanating from a source closer to one of the microphones than another of the microphones; and wherein the processing includes determining the signal weights as a function of an amplitude difference between the signals for each of a number of different frequencies in the frequency domain. 28. A sound processing apparatus, comprising: an 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 processor operable to define an adaptive beamformer with the sensor signals and generate an output signal representative of sound emanating from a selected source, the processor being responsive to a sound interference level to adjust beamwidth of the beamformer.
연구과제 타임라인
LOADING...
LOADING...
LOADING...
LOADING...
LOADING...
이 특허에 인용된 특허 (102)
Funke Hermann D. (Bonn DEX), Acoustic body bus medical device communication system.
Sejnowski Terrence (Solana Beach CA) Li Shao L. (San Diego CA), Adaptive system for broadband multisignal discrimination in a channel with reverberation.
Bond James W. (San Diego CA) Alsup James M. (San Diego CA) Speiser Jeffrey M. (San Diego CA) Whitehouse Harper J. (San Diego CA) Lagnado Isaac (San Diego CA), CCD Analog and digital correlators.
Kroll Mark W. (Minnetonka MN) Adams Theodore P. (Edina MN) Brumwell Dennis A. (Bloomington MN), Implantable defibrillator system with capacitor switching circuitry.
Franklin David (Somerville MA), Method and apparatus for reducing background noise in communication systems and for enhancing binaural hearing systems f.
Soli Sigfrid D. (Sierra Madre CA) Jayaraman Sriram (Los Angeles CA) Gao Shawn (Cerritos CA) Sullivan Jean (Murrieta CA), Method of signal processing for maintaining directional hearing with hearing aids.
Brandstein Michael S. ; Adcock John E. ; Silverman Harvey F., Methods and apparatus for source location estimation from microphone-array time-delay estimates.
White Stanley A. ; Walley Kenneth S. ; Johnston James W. ; Henderson P. Michael ; Hale Kelly H. ; Andrews ; Jr. Warner B. ; Siann Jonathan I., System and method for a monolithic directional microphone array.
Shannon Dorothy A. (Ellicott City MD) King John P. (Ellicott City MD) DeWitte ; Jr. Joseph T. (Laurel MD) Orndorff James D. (Baltimore MD), Ultrasonic frequency expansion processor.
※ AI-Helper는 부적절한 답변을 할 수 있습니다.