Methods and apparatus for mitigating the effects of microphone overload in echo cancelation systems

• US 6,850,783 B1
• Filed: 08/07/1998
• Issued: 02/01/2005
• Est. Priority Date: 08/07/1998
• Status: Expired due to Fees

• Current Assignee: Ericsson, Inc. (Telefonaktiebolaget LM Ericsson)
• Original Assignee: Ericsson, Inc. (Telefonaktiebolaget LM Ericsson)
• Inventors: Romesburg, Eric Douglas
• Primary Examiner(s): Corsaro, Nick
• Application Number:
  US09/131,167
• Time in Patent Office: 2,370 Days
• Field of Search: 455/569, 455/570, 455/569.1, 379/420, 379/406, 379/410, 379/411, 379/406.04, 381/96, 381/95, 381/93
• US Class Current: 455/570
• CPC Class Codes: H04M 9/082 using echo cancellers echo cancellers per se H04B3/23

Abstract

Methods and apparatus for echo and noise cancelation contemplate occasional saturation of an echo and/or noise containing source signal and control adaptation of a cancelation filter to minimize the detrimental effects of such source signal saturation. According to exemplary embodiments, filter coefficients of an adaptive filter are updated only when a prevailing value of the source signal is within a predetermined range of possible source signal values. Consequently, the filter response is not perturbed by source signal saturation, and the adaptive filter continues to provide quality echo suppression even immediately following periods of saturation.

Claims

1. A communications device, comprising:

a microphone;

a loudspeaker;

an adaptive filter configured to estimate an echo component of an output signal of said microphone based on an input signal of said loudspeaker; and

a summing device configured to remove the echo estimate from the output signal of the microphone and to thereby provide an echo canceled output signal, wherein the adaptive filter updates filter coefficients only when a prevailing value of the output signal of said microphone is within a predetermined range of possible output signal values less than a total range of possible output signal values.

2. A communications device according to claim 1, wherein the filter coefficients of said adaptive filter are updated only when the prevailing value of the output signal of said microphone is below a predetermined threshold value.

3. A communications device according to claim 1, wherein the filter coefficients of said adaptive filter are updated only when the prevailing value of the output signal of said microphone is above a predetermined threshold value.

4. A communications device according to claim 1, wherein the filter coefficients of said adaptive filter are updated only when the prevailing value of the output signal of said microphone is below a first predetermined threshold value and above a second predetermined threshold value.

5. A communications device according to claim 4, further comprising one of a sampling analog-to-digital converter and a Σ

-Δ

analog-to-digital converter, said one analog-to-digital converter being configured to digitize the output signal of said microphone, wherein said first predetermined threshold value is less than or equal to an upper full-scale value of said analog-to-digital converter, and wherein said second predetermined threshold value is greater than or equal to a lower full-scale value of said analog-to-digital converter.

6. A communications device according to claim 4, wherein at least one of the first and second predetermined threshold values is adjusted to account for analog distortion.

7. A communications device according to claim 4, wherein said one analog-to-digital converter is a Σ

-Δ

converter and wherein at least one of the predetermined threshold values is adjusted to account for ringing of said Σ

-Δ

converter.

8. A communications device according to claim 1, wherein said communications device is configured to operate as a mobile telephone.

9. A communications device according to claim 1, wherein the filter coefficients of said adaptive filter are updated according to a Least Mean Squares (LMS) algorithm.

10. A communications device according to claim 1, wherein the filter coefficients of said adaptive filter are updated according to a Normalized Least Mean Squares (NLMS) algorithm.

11. A communications device according to claim 1, wherein the filter coefficients are updated in a sample-wise fashion and wherein a sample-wise update of the filter coefficients is performed only when a value of a corresponding sample of the output signal of said microphone is within a predetermined range of possible values.

12. A communications device according to claim 1, wherein the filter coefficients are updated in a blockwise fashion and wherein a blockwise update of the filter coefficients is performed only when a value of each one of a number of corresponding samples of the output signal of said microphone is within a predetermined range of possible output signal values.

13. An echo canceler for use in a communications device including a microphone and a loudspeaker, said echo canceler comprising:

an adaptive filter configured to estimate an echo component of an output signal of the microphone based on an input signal of the loudspeaker; and

a summing device configured to remove the echo estimate from the output signal of the microphone and to thereby provide an echo canceled output signal,

wherein the adaptive filter updates filter coefficients only when a prevailing value of the output signal of the microphone is within a predetermined range of possible output signal values less than a total range of possible output signal values.

14. An echo canceler according to claim 13, wherein the filter coefficients of said adaptive filter are updated only when the prevailing value of the output signal of the microphone is below a predetermined threshold value.

15. An echo canceler according to claim 13, wherein the filter coefficients of said adaptive filter are updated only when the prevailing value of the output signal of the microphone is above a predetermined threshold value.

16. An echo canceler according to claim 13, wherein the filter coefficients of said adaptive filter are updated only when the prevailing value of the output signal of the microphone is below a first predetermined threshold value and above a second predetermined threshold value.

17. A communications device, comprising:

an adaptive system configured to estimate a component of a source signal, wherein the adaptive system updates variables only when a prevailing value of the source signal is within a predetermined range of possible source signal values less than a total range of possible output signal values.

18. A communications device according to claim 17, wherein the component of the source signal is an echo component.

19. A communications device according to claim 17, wherein the component of the source signal is a noise component.

20. A communications device according to claim 17, further comprising a device for removing the component estimate from the source signal and to thereby provide a component-canceled output signal.

21. A communications device according to claim 17, wherein said source signal is a microphone output signal, wherein said component estimate is computed based on a loudspeaker input signal, and wherein said communications device is configured to operate as an acoustic echo canceler.

22. A communications device according to claim 17, wherein said source signal is a far-end communications signal, wherein said component estimate is computed based on a near-end communications signal, and wherein said communications device is configured to operate as a network echo canceler.

23. A communications device according to claim 17, wherein said communications device is configured to operate as a noise suppressor.

24. A communications device according to claim 17, wherein said communications device is configured to operate as spectral subtraction processor.

25. A method for canceling and echo component of a communications signal, the echo component resulting from and echo-causing signal, said method comprising the steps of:

filtering the echo causing signal using an adaptive filter to provide an estimate of the echo component of the communications signal;

subtracting the echo estimate from the communications signal to provide an echo canceled output signal; and

updating filter coefficients of the adaptive filter, based on the echo canceled output signal and the echo causing signal, only when a prevailing value of the communications signal is within a predetermined range of possible signal values less than a total range of possible output signal values.

26. The method of claim 25, wherein said step of updating filter coefficients of the adaptive filter comprises the steps of:

comparing the prevailing value of the communications signal with a predetermined threshold value; and

updating the filter coefficients only if the prevailing value is less than the predetermined threshold value.

27. The method of claim 25, wherein said step of updating filter coefficients of the adaptive filter comprises the steps of:

comparing the prevailing value of the communications signal with a predetermined threshold value; and

updating the filter coefficients only if the prevailing value is greater than the predetermined threshold value.

28. The method of claim 25, wherein said step of updating filter coefficients of the adaptive filter comprises the steps of:

comparing the prevailing value of the communications signal with first and second predetermined threshold values; and

updating the filter coefficients only if the prevailing value is less than the first predetermined threshold value and greater than the second predetermined threshold value.

29. The method of claim 28, wherein one of a sampling analog-to-digital converter and a Σ

-Δ

analog-to-digital converter is used to digitize the output signal of said microphone, wherein the first predetermined threshold value is less than or equal to a upper full-scale value of said analog-to-digital converter, and wherein the second predetermined threshold value is greater than or equal to a lower full-scale value of said analog-to-digital converter.

30. The method of claim 29, wherein at least one of the predetermined threshold values is adjusted to account for analog compression.

31. The method of claim 29, wherein said one analog-to-digital converter is a Σ

-Δ

converter and wherein at least one of the predetermined threshold values is adjusted to account for ringing of said converter.

32. The method of claim 25, wherein the communications signal is a microphone signal and wherein the echo causing signal is a loudspeaker signal.

33. The method of claim 25, wherein the communications signal is a near-end speech signal and wherein the echo causing signal is a far-end speech signal.

34. The method of claim 25, wherein the filter coefficients of the adaptive filter are updated according to a Least Mean Square (LMS) algorithm.

35. The method of claim 25, wherein the filter coefficients of the adaptive filter are updated according to a Normalized Least Mean Square (NLMS) algorithm.

36. The method of claim 25, wherein the filter coefficients are updated in a sample-wise fashion and wherein a sample-wise update of the filter coefficients is performed only when a value of a corresponding sample of the communications signal is within a predetermined range of possible signal values.

37. The method of claim 25, wherein the filter coefficients are updated in a blockwise fashion and wherein a blockwise update of the filter coefficients is performed only when a value of each one of a number of corresponding samples of the communications signal is within a predetermined range of possible signal values.

38. An echo canceler for use in a communications device including a microphone and a loudspeaker, said echo canceler comprising:

an adaptive filter that estimates an echo component of an output signal of the microphone based on an input signal of the loudspeaker;

means for updating filter coefficients of said adaptive filter only when an absolute value of the output signal of the microphone is less than a saturation constant; and

a summing device that removes the echo estimate from the output signal of the microphone, thereby providing an echo canceled output signal.

39. The communications device of claim 38, wherein the saturation constant is based on at least one component used in a signal path of the microphone output.

40. The communications device of claim 39, wherein the at least one component is a sampling type analog-to-digital converter that is used to digitize the microphone output signal, and wherein the saturation constant is set at about a full scale value of an input range of said analog-to-digital converter.

41. The communications device of claim 39, wherein the at least one component is a Σ

-Δ

type analog-to-digital converter that is used to digitize the microphone output signal, and wherein the saturation constant is set below a full scale value of an input range of said analog-to-digital converter to account for ringing of said converter.