[특허]Updating modeling information based on online data gathering

[미국특허] Updating modeling information based on online data gathering 원문보기

IPC분류정보
국가/구분	United States(US) Patent 등록
국제특허분류(IPC7판)	H04B-003/20 H04R-003/00
출원번호	UP-0405683 (2006-04-17)
등록번호	US-7760887 (2010-08-09)
발명자 / 주소	Oxford, William V.
출원인 / 주소	LifeSize Communications, Inc.
대리인 / 주소	Meyertons Hood Kivlin Kowert & Goetzel, P.C.
인용정보	피인용 횟수 : 3 인용 특허 : 73

초록 ▼

A system may include a processor and memory. The processor may be configured to perform calibration measurements on the speaker even when the speaker is being used to conduct a live conversation. The processor may be configured to: provide a live output signal for transmission from a speaker; receive an input signal corresponding to the output signal; compute 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; subtract the midrange sensitivity from the lowpass sensitivity to obtain a speaker-related sensitivity; perform an iterative search for current parameters of a speaker model using the input signal spectrum, the output signal spectrum and the speaker-related sensitivity; and update averages of the speaker model parameters using the current parameter values. The parameter averages may be used to perform echo cancellation.

대표청구항 ▼

What is claimed is: 1. A method comprising: (a) providing an output signal for transmission from a speaker, wherein the output signal carries live signal information from a remote source; (b) receiving an input signal from a microphone; (c) computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of a portion of the input signal and a spectrum of a portion of the output 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 spectrum of the input signal portion, the spectrum of the output signal portion, and 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 further comprising: computing an average power signal on a stream of samples of the live signal information; determining a window in time when the average power signal has remained greater than a power threshold for a predetermined amount of time; wherein said portion of the output signal corresponding to samples of the live signal information during said window in time. 3. The method of claim 1, wherein the portion of the input signal corresponds to the portion of the output signal and a reverb tail of the portion of the output signal. 4. The method of claim 1 further comprising: storing a plurality of portions of the output signal and corresponding portions of the input signal; performing (c) through (f) a number of times, wherein each iteration of performing (c) through (f) operates on one of the output signal portions and the corresponding input signal portion. 5. The method of claim 1 further comprising: comparing the updated parameter averages respectively to previous values of the parameter averages; generating a problem report for the speaker in response to a determination that one or more of the updated parameter averages departs by more than a corresponding predetermined amount from the corresponding previous value. 6. The method of claim 1 further comprising: applying one or more notch filters to the output signal prior to transmission from the speaker in order to remove one or more frequencies from the output signal. 7. The method of claim 1, wherein said updating the parameter averages using the current parameter values is performed according to an infinite impulse response (IIR) filtering scheme. 8. The method of claim 1 further comprising: performing an iterative search for a current transfer function of the microphone using the spectrum of the input signal portion, the spectrum of the output signal portion, and the updated parameter averages; updating an average microphone transfer function using the current microphone transfer function; wherein the average microphone transfer function is usable to perform echo cancellation. 9. The method of claim 1, wherein (a) through (f) are performed by one or more processors in a speakerphone. 10. The method of claim 1, wherein the parameter averages of the speaker input-output model are usable to perform echo cancellation on the input signal. 11. The method of claim 1, wherein the input-output model of the speaker is a nonlinear model. 12. The method of claim 3, wherein the nonlinear model is a Volterra series model. 13. A computer-readable memory medium configured to store program instructions, wherein the program instructions are executable to implement: (a) providing an output signal for transmission from a speaker, wherein the output signal carries live signal information from a remote source; (b) receiving an input signal from a microphone; (c) computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of a portion of the input signal and a spectrum of a portion of the output 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 spectrum of the input signal portion, the spectrum of the output signal portion, and the speaker-related sensitivity; and (f) updating averages of the parameters of the speaker input-output model using the current parameter values. 14. The method of claim 13, wherein the program instructions are executable to further implement: computing an average power signal on a stream of samples of the live signal information; determining a window in time when the average power signal has remained greater than a power threshold for a predetermined amount of time; wherein said portion of the output signal corresponding to samples of the live signal information during said window in time. 15. The method of claim 13, wherein the portion of the input signal corresponds to the portion of the output signal and a reverb tail of the portion of the output signal. 16. The method of claim 13, wherein the program instructions are executable to further implement: storing a plurality of portions of the output signal and corresponding portions of the input signal; performing (c) through (f) a number of times, wherein each iteration of performing (c) through (f) operates on one of the output signal portions and the corresponding input signal portion. 17. The method of claim 13, wherein the program instructions are executable to further implement: comparing the updated parameter averages respectively to previous values of the parameter averages; generating a problem report for the speaker in response to a determination that one or more of the updated parameter averages departs by more than a corresponding predetermined amount from the corresponding previous value. 18. 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) providing an output signal for transmission from a speaker, wherein the output signal carries live signal information from a remote source; (b) receiving an input signal from a microphone; (c) computing a midrange sensitivity and a lowpass sensitivity for a transfer function derived from a spectrum of a portion of the input signal and a spectrum of a portion of the output 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 spectrum of the input signal portion, the spectrum of the output signal portion, and the speaker-related sensitivity; and (f) updating averages of the parameters of the speaker input-output model using the current parameter values. 19. The system of claim 18 further comprising the speaker and the microphone. 20. The system of claim 18, wherein the updated parameter averages are usable to perform echo cancellation on the input signal.

이 특허에 인용된 특허 (73) 인용/피인용 타임라인 분석

Mitchell A. Markow ; Jeremy Ford ; Ji-An Gong, Acoustic coupling compensation for a speakerphone of a system.
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Berstein, Alberto D.; Wei, Abraham; Wei, Tom, Acoustic echo processing system.
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Chu Peter L. (Needham MA), Adaptive acoustic echo canceller having means for reducing or eliminating echo in a plurality of signal bandwidths.
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David V. Anderson ; Stephen McGrath ; Kwan Truong, Adaptive filter featuring spectral gain smoothing and variable noise multiplier for noise reduction, and method therefor.
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Buchele, William N., Adaptive speech filter.
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O'Malley, William; Leondires, Arthur P., Audio conference platform system and method for broadcasting a real-time audio conference over the internet.
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O'Malley, William; Grandgent, Charles M., Audio conference platform with dynamic speech detection threshold.
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Stautner John P. (Wellesley Hills MA) Morrell William R. (Somerville MA) Jayasimha Sriram (Boston MA), Audio decompression system employing multi-rate signal analysis.
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Botzko Stephen C. ; Franklin David M., Audio processor.
상세보기
Botzko Stephen C. ; Franklin David M., Audio processor.
상세보기
McCaskill, John, Automatic camera tracking using beamforming.
상세보기
Epps David C. (Denver CO), Automatic gain control using root-mean-square circuitry in a digital domain conference bridge for a telephone network.
상세보기
Graumann David L., Automatic microphone calibration for video teleconferencing.
상세보기
Maeng, Joonyoul; Williams, Errol R., Automatic voice tracking camera system and method of operation.
상세보기
Crow Robert P. (4725 Bywood Ct. Colorado Springs CO 80906), Beamed linear array microphone system.
상세보기
Mller Bernhard (Berlin DEX), Boundary-layer microphone.
상세보기
Epps David C. (Denver CO), Circuit for nulling the talker\s speech in a conference call and method thereof.
상세보기
Faraci Philip ; Skene Jeremy F., Compact speakerphone apparatus.
상세보기
Kang, Kyeong Ok; Jang, Dae Young; Kwak, Jin Suk; Hong, Jin Woo; Kim, Sung Han; Lim, Young Kwon; Kim, Jin Woong; Popp, Harald; Geyersberger, Stefan; Fiesel, Wolfgang, Data CODEC system for computer.
상세보기
Ludwig, Lester F.; Lauwers, J. Chris; Lantz, Keith A.; Burnett, Gerald J.; Burns, Emmett R., Dedicated echo canceler for a workstation.
상세보기
Van Gerwen Petrus J. (Eindhoven NLX) van de Laar Franciscus A. M. (Eindhoven NLX) Kotmans Hendrik J. (Eindhoven NLX), Digital echo canceller.
상세보기
Paul Jon D. (2800 Third St. San Francisco CA 94107) Clayton Mark D. (Box 541 ; RD 4 New Windsor NY 12550) Agnello Anthony M. (407 Prospect Ave. Princeton NJ 08540), Digital output transducer.
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Takaaki Kishigami JP; Takashi Fukagawa JP; Makoto Hasegawa JP, Direction of arrival estimation apparatus and variable directional signal receiving and transmitting apparatus using the same.
상세보기
Stewart, Brett B., Directional microphone utilizing spaced apart omni-directional microphones.
상세보기
Furuta,Satoru; Takahashi,Shinya, Echo processor generating pseudo background noise with high naturalness.
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Castles,Randy; Sexton,Martin; Nimri,Alain, Graphical user interface for video feed on videoconference terminal.
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Carver, Robert W., High back-EMF high pressure subwoofer having small volume cabinet low frequency cutoff and pressure resistant surround.
상세보기
Baxley, Warren E.; Clemson, Gregory A.; Ernstrom, William P.; Nylander, Eric J.; Stark, Thomas W.; Yackey, Jr., Thomas E., Large-scale, fault-tolerant audio conferencing in a purely packet-switched network.
상세보기
Baxley, Warren E.; Clemson, Gregory A.; Ernstrom, William P.; Nylander, Eric J.; Stark, Thomas W.; Yackey, Jr., Thomas E., Large-scale, fault-tolerant audio conferencing over a hybrid network.
상세보기
Potts, Steven L.; Wang, Hong; Rabiner, Wendi Beth; Chu, Peter L., Locating an audio source.
상세보기
Randmere Uno (Victor NY) Suntop Morris A. (Rochester NY), Loudspeaking telephone hysteresis and ambient noise control.
상세보기
Baumhauer ; Jr. John C. (Indianapolis IN) McAteer Jeffrey P. (Fishers IN) Wittman Brian A. (Indianapolis IN), Loudspeaking telephone station including directional microphone.
상세보기
Chu Peter L. (Needham MA), Method and apparatus for estimating the level of acoustic feedback between a loudspeaker and microphone.
상세보기
Chu Peter L. ; Wang Hong, Method and apparatus for localization of an acoustic source.
상세보기
Hinman Brian L. (Los Gatos CA) Huang Shan-Shan (Sunnyvale CA) Gaut Eric K. (Foster City CA), Method and apparatus for periodic signal detection.
상세보기
Aoki Mariko,JPX ; Aoki Shigeaki,JPX ; Matsui Hiroyuki,JPX ; Nishino Yutaka,JPX ; Okamoto Manabu,JPX, Method and apparatus for separation of source, program recorded medium therefor, method and apparatus for detection of sound source zone, and program recorded medium therefor.
상세보기
Chu Peter L., Method and apparatus for steerable and endfire superdirective microphone arrays with reduced analog-to-digital converter.
상세보기
Strubbe, Hugo J.; Lee, Mi-Suen, Method and apparatus for tracking moving objects using combined video and audio information in video conferencing and other applications.
상세보기
Huang Shan-Shan (Sunnyvale CA) Hinman Brian L. (Los Gatos CA) Gaut Eric K. (Palo Alto CA), Method for fail-safe operation in a speaker phone system.
상세보기
Dominique Pastor NL; Gerard Reynaud FR; Pierre-Albert Breton FR, Method of frequency filtering applied to noise suppression in signals implementing a wiener filter.
상세보기
Brandstein Michael S. (Providence RI) Silverman Harvey F. (East Greenwich RI), Methods and apparatus for adaptive beamforming.
상세보기
Brandstein Michael S. ; Adcock John E. ; Silverman Harvey F., Methods and apparatus for source location estimation from microphone-array time-delay estimates.
상세보기
Ludwig ; Jr. Lester Frank, Methods and systems for multimedia communications via public telephone networks.
상세보기
Sudo, Yoshinori; Fujiwara, Noritaka, Microphone holder having connector unit molded together with conductive strips.
상세보기
Chu Peter Lee (Lexington MA) Barton William F. (Littleton MA), Microphone system for teleconferencing system.
상세보기
Chu Peter Lee ; Barton William F., Microphone system for teleconferencing system.
상세보기
He,Perry P.; Smith,Roger A.; Pessoa,Lucio F. C.; Dyba,Roman A., Monitoring and control of an adaptive filter in a communication system.
상세보기
Hirano Akihiro (Tokyo JPX) Sugiyama Akihiko (Tokyo JPX), Multi-channel echo canceling method and apparatus.
상세보기
Ludwig Lester F. (Foster City CA) Lauwers J. Chris (Menlo Park CA) Lantz Keith A. (Los Altos CA) Burnett Gerald J. (Atherton CA) Burns Emmett R. (Incline Village NV), Multimedia collaboration system arrangement for routing compressed AV signal through a participant site without decompre.
상세보기
Ludwig Lester F. (Foster City CA) Lauwers J. Chris (Menlo Park CA) Lantz Keith A. (Los Altos CA) Burnett Gerald J. (Atherton CA) Burns Emmett R. (Incline Village NV), Multimedia collaboration system with separate data network and A/V network controlled by information transmitting on the.
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Ludwig, Lester; Brown, William Blake; Yul, Inn J.; Vuong, Anh T.; Vanderlippe, Richard W.; Burnett, Gerald; Lauwers, Chris; Lui, Richard; Applebaum, Daniel, Networked video multimedia storage server environment.
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Tucker, Mike; Binford, Don; Flott, Rick; Vanderwilt, Patrick; Kenoyer, Michael; Weaver, Jason C.; Manning, Earl, Personal videoconferencing system having distributed processing architecture.
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Hirai,Toru; Kawakami,Fukushi; Shimizu,Yasushi; Honji,Yoshikazu; Ikeda,Masahiro; Tohyama,Mikio, Picked-up-sound recording method and apparatus.
상세보기
Anil Wamanrao Ubale ; Grant Allen Davidson, Quantization in perceptual audio coders with compensation for synthesis filter noise spreading.
상세보기
Helf Brant M. (Melrose MA) Chu Peter L. (Lexington MA), Reduction of background noise for speech enhancement.
상세보기
Donovan, Steven R.; Devanathan, Raghavan, SIP-based feature control.
상세보기
O'Malley, William; Leondires, Arthur P., Scalable audio conference platform.
상세보기
Yoshino,Hajime, Sound characteristic measuring device, automatic sound field correcting device, sound characteristic measuring method and automatic sound field correcting method.
상세보기
Curry,James E.; McAllister,Alexander I.; Hatton,Patricia V., Spatial sound conference system and apparatus.
상세보기
David V. Anderson ; Stephen McGrath ; Kwan Truong, Speech activity detector for use in noise reduction system, and methods therefor.
상세보기
Ding, Heping, Stable adaptive filter and method.
상세보기
Duttweiler Donald L. (Rumson NJ), Subband echo canceler.
상세보기
Chu, Peter L.; Kenoyer, Michael; Washington, Richard, System and method for computing a location of an acoustic source.
상세보기
Shachar, Yair; Winter, Isac; Menczel, Yaron, System and method for videoconference initiation.
상세보기
Deierling Kevin, System and method for videoconferencing and simultaneously viewing a supplemental video source.
상세보기
Huang Jixiong ; Grinnell Richard S., Teleconferencing system with visual feedback.
상세보기
Davidson Grant Allen ; Robinson Charles Quito ; Truman Michael Mead, Using gain-adaptive quantization and non-uniform symbol lengths for improved audio coding.
상세보기
Helf Brant, Variable length coding using a plurality of region bit allocation patterns.
상세보기
Ippolito Peter M. ; Cook Caroline M., Video conferencing apparatus for group video conferencing.
상세보기
Gupta Prabhat K. (Germantown MD) Jangi Shrirang (Germantown MD) Lamkin Allan B. (Arlington VA) Kepley ; III W. Robert (Gaithersburg MD) Morris Adrian J. (Gaithersburg MD), Voice activity detector for speech signals in variable background noise.
상세보기
Beecroft, Troy C., Voice telephony system.
상세보기
Provencal, Paul; Forrester, Christopher M, Voice-switching device and method for multiple receivers.
상세보기
Epps, David C., Whisper circuit for a conference call bridge including talker nulling and method therefor.
상세보기

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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 online data gathering 원문보기

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