[특허]Updating modeling information based on offline calibration experiments

[미국특허] Updating modeling information based on offline calibration experiments 원문보기

IPC분류정보
국가/구분	United States(US) Patent 등록
국제특허분류(IPC7판)	H04R-003/00 H04B-003/20
출원번호	UP-0405667 (2006-04-14)
등록번호	US-7720236 (2010-06-10)
발명자 / 주소	Oxford, William V.
출원인 / 주소	LifeSize Communications, Inc.
대리인 / 주소	Meyertons Hood Kivlin Kowert & Goetzel, P.C.
인용정보	피인용 횟수 : 2 인용 특허 : 73

초록 ▼

A system such as a speakerphone may include a processor, memory, a speaker and a microphone. The processor may be configured (via program instructions stored in the memory) to calibrate the speaker by: outputting a stimulus signal; receiving an input signal corresponding to the stimulus signal; computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of the input signal and a spectrum of the output signal; subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; performing an iterative search for current parameters of a speaker model using the input signal spectrum, the stimulus signal spectrum and the speaker-related sensitivity; and updating averages of the speaker model parameters using the current parameter values. The stimulus signal may be transmitted during periods of silence in the external environment. The parameter averages may be used to perform echo cancellation.

대표청구항 ▼

What is claimed is: 1. A method comprising: (a) outputting a stimulus signal for transmission from a speaker; (b) receiving an input signal from a microphone, wherein the input signal corresponds to the stimulus signal; (c) computing a midrange sensitivity and a lowpass sensitivity for a transfer function H(ω) derived from a spectrum of the input signal and a spectrum of the stimulus signal; (d) subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; (e) performing an iterative search for current values of parameters of an input-output model of the speaker using the input signal spectrum, the stimulus signal spectrum, the speaker-related sensitivity; and (f) updating averages of the parameters of the speaker input-output model using the current parameter values. 2. The method of claim 1, wherein the parameter averages of the speaker input-output model are usable to perform echo cancellation on second input signals from the microphone. 3. The method of claim 1, wherein the input-output model of the speaker is a nonlinear model. 4. The method of claim 3, wherein the nonlinear model is a Volterra series model. 5. The method of claim 1, wherein the stimulus signal is a noise signal. 6. The method of claim 5, wherein the noise signal is maximum length sequence noise. 7. The method of claim 1 further comprising: applying one or more notch filters to the stimulus signal prior to transmission from the speaker in order to remove one or more frequencies from the stimulus signal, wherein the one or more frequencies are known to induce resonance in one or more physical structures. 8. The method of claim 1 further comprising: performing an iterative search for a current transfer function of the microphone using the input signal spectrum, the spectrum of the stimulus signal, and the current parameter values; updating an average microphone transfer function using the current transfer function; wherein the average transfer function is usable to perform said echo cancellation. 9. The method of claim 1 further comprising monitoring average signal power from the microphone, wherein said outputting of the stimulus signal is performed in response to a determination that the average signal power from the microphone has remained less than a power threshold for a predetermined amount of time. 10. The method of claim 1, wherein (a) through (f) are performed by one or more processors in a speakerphone. 11. A computer-readable memory medium configured to stored program instructions, wherein the program instructions are executable to implement: (a) receiving an input signal from a microphone, wherein the input signal corresponds to a transmission of a stimulus signal from a speaker; (b) computing a midrange sensitivity and a lowpass sensitivity for a transfer function H(ω) derived from a spectrum of the input signal and a spectrum of the stimulus signal; (c) subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; (d) performing an iterative search for current values of parameters of an input-output model of the speaker using the input signal spectrum, the stimulus signal spectrum, the speaker-related sensitivity; and (e) updating averages of the parameters of the speaker input-output model using the current parameter values. 12. The method of claim 11, wherein the parameter averages of the speaker input-output model are usable to perform echo cancellation on second input signals received from the microphone. 13. The method of claim 11, wherein the input-output model of the speaker is a nonlinear model. 14. The method of claim 13, wherein the nonlinear model is a Volterra series model. 15. The method of claim 11, wherein the stimulus signal is a noise signal. 16. The method of claim 11 further comprising: applying one or more notch filters to the stimulus signal prior to transmission from the speaker in order to remove one or more frequencies from the stimulus signal, wherein the one or more frequencies are known to induce resonance in one or more physical structures. 17. The method of claim 11 further comprising: performing an iterative search for a current transfer function of the microphone using the input signal spectrum, the spectrum of the stimulus signal, and the current parameter values; updating an average microphone transfer function using the current transfer function; wherein the average transfer function is usable to perform said echo cancellation. 18. The memory medium of claim 11, wherein the program instructions are further executable to implement: monitoring average signal power from the microphone, wherein said outputting of the stimulus signal is performed in response to a determination that the average signal power from the microphone has remained less than a power threshold for a predetermined amount of time. 19. A system comprising: a memory configured to store program instructions; a processor configured to read and execute the program instructions from the memory, wherein the program instructions are executable by the processor to implement: (a) receiving an input signal from a microphone, wherein the input signal corresponds to a transmission of a stimulus signal from a speaker; (b) computing a midrange sensitivity and a lowpass sensitivity for a transfer function H(ω) derived from a spectrum of the input signal and a spectrum of the stimulus signal; (c) subtracting the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; (d) performing an iterative search for current values of parameters of an input-output model of the speaker using the input signal spectrum, the stimulus signal spectrum, the speaker-related sensitivity; and (e) updating averages of the parameters of the speaker input-output model using the current parameter values. 20. The system of claim 19 further comprising the speaker and the microphone.

이 특허에 인용된 특허 (73) 인용/피인용 타임라인 분석

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Gygax, Otto A.; Hu, Deqing; Davis, Joseph, Integrating acoustic echo cancellation as a subsystem into an existing videoconference and telepresence system.
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[미국특허] Updating modeling information based on offline calibration experiments 원문보기

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