Updating modeling information based on offline calibration experiments

• US 7,720,236 B2
• Filed: 04/14/2006
• Issued: 05/18/2010
• Est. Priority Date: 10/15/2004
• Status: Active Grant

• Current Assignee: Lifesize, Inc. (Enghouse Systems Limited)
• Original Assignee: Lifesize, Inc. (Enghouse Systems Limited)
• Inventors: Oxford, William V.
• Primary Examiner(s): Mei; Xu
• Application Number:
  US11/405,667
• Publication Number:
  US 20060269074A1
• Time in Patent Office: 1,495 Days
• Field of Search: 381/59, 381/61, 381/66, 381/91, 381/92, 381/95, 381/96, 381/98, 381/122, 379/406.01, 379/406.02, 379/406.06, 37940612-40615
• US Class Current: 381/96
• CPC Class Codes: H04M 1/6008 in the transmitter circuit
• CPC Class Codes: H04M 1/6033 for providing handsfree use or a loudspeaker mode in telephone sets arrangements for preventing acoustic echo H04M9/08
• CPC Class Codes: H04R 1/403 loud-speakers
• CPC Class Codes: H04R 1/406 microphones
• CPC Class Codes: H04R 2201/401 2D or 3D arrays of transducers
• CPC Class Codes: H04R 2201/403 Linear arrays of transducers
• CPC Class Codes: H04R 2201/405 Non-uniform arrays of transducers or a plurality of uniform arrays with different transducer spacing
• CPC Class Codes: H04R 2203/12 Beamforming aspects for stereophonic sound reproduction with loudspeaker arrays
• CPC Class Codes: H04R 2225/41 Detection or adaptation of hearing aid parameters or programs to listening situation, e.g. pub, forest
• CPC Class Codes: H04R 2227/003 Digital PA systems using, e.g. LAN or internet
• CPC Class Codes: H04R 2420/07 Applications of wireless loudspeakers or wireless microphones
• 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 2430/21 Direction finding using differential microphone array [DMA]
• CPC Class Codes: H04R 2430/23 Direction finding using a sum-delay beam-former
• CPC Class Codes: H04R 2499/11 Transducers incorporated or for use in hand-held devices, e.g. mobile phones, PDA's, camera's
• CPC Class Codes: H04R 27/00 Public address systems circuits for preventing acoustic reaction H04R3/02; circuits for distributing signals to loudspeakers H04R3/12; monitoring or testing arrangements for public address systems H04R29/007; amplifiers H03F
• CPC Class Codes: H04R 29/00 Monitoring arrangements; Testing arrangements for hearing aids H04R25/30; detection of loudspeaker connection H04R5/04; sound-field adaptation dependent on speaker detection H04S7/308
• CPC Class Codes: H04R 29/007 for public address systems public address systems per se H04R27/00
• CPC Class Codes: H04R 3/005 for combining the signals of two or more microphones specially adapted for hearing aids H04R25/407
• CPC Class Codes: H04R 3/02 for preventing acoustic reaction , i.e. acoustic oscillatory feedback specially adapted for hearing aids H04R25/453

Abstract

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.

Claims

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.