IP Library Granted Patent US 8,675,884
Granted Patent B2
US 8,675,884 · App. 12/990,647 · Granted Mar 18, 2014

Method and a system for processing signals

Inventors: Uri Yehuday (Bat Yam, IL); Arie Heiman (Raanana, IL)
Assignee: DSP Group
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Quick Facts
Patent No.
US 8,675,884
App. No.
12/990,647
Granted
Mar 18, 2014
Kind
B2
Abstract

A system for processing sound, the system including: (a) a processor, configured to process a first input signal that is detected by a first microphone at a detection moment, a second input signal that is detected by a second microphone at the detection moment, and a third input signal that is detected by a bone-conduction microphone at the detection moment, to generate a corrected signal that is responsive to the first, second, and third input signals; and (b) a communication interface, configured to provide the corrected signal to an external system.

Claims (38)

1. A system for processing sound, the system comprising:

a processor, configured to process a first input signal that is detected by a first microphone at a detection moment, a second input signal that is detected by a second microphone at the detection moment, and a third input signal that is detected by a bone-conduction microphone at the detection moment, to generate a corrected signal that is responsive to the first, second, and third input signals; and a communication interface, configured to provide the corrected signal to an external system; wherein the processor is further configured to determine the corrected signal Ŝ(n) for the detection moment n, by a sum of convolutions Ŝ(n)=h 1 (n)*M 1 (n)+h 2 (n)*M 2 (n)+h 3 (n)*M 3 (n), wherein M 1 (n) represents the first input signal at the detection moment, M 2 (n) represents the second input signal at the detection moment, M 3 (n) represents the third input signal at the detection moment, and h 1 (n), h 2 (n), and h 3 (n) are calibration functions.

2. The system of claim 1 , wherein the second input signal is detected by the second microphone which is placed at least partly within an ear of a user.

3. The system of claim 2 , wherein the second input signal is responsive to a sound signal which was modified within the ear canal, so that lower frequencies of the sound signal were amplified within the ear canal.

4. The system of claim 2 , wherein the second microphone is comprised in an ear plug that blocks the ear canal to ambient sound.

5. The system of claim 1 , wherein the processor is further configured to update at least one calibration function in response to processing of input signals at a past moment that precedes the detection moment.

6. The system of claim 5 , wherein the processor is configured to selectively update the at least one calibration function for at least one past moment in which a speaking of a user is detected.

7. The system of claim 6 , wherein the processor is further configured to detect a speaking of a user in the past moment by analyzing a speaking spectrum of at least one input signal.

8. The system of claim 5 , wherein the processor is configured to update the at least one calibration function in response to an error function {tilde over (e)}(n) the value of which for the detection moment n is determined by {tilde over (e)}(n)≈{circumflex over (γ)}(n)*{tilde over (s)}(n)−M 3 (n) where {tilde over (s)}(n) is a sum of H 1 (z), H 2 (z), and H 3 (z), wherein H i (z) is the Z-transform of the corresponding calibration function h i (n).

9. The system of claim 5 , wherein the processor is further configured to update a calibration function h i (n) is responsive to a partial derivative of a mean square error function J with respect to the calibration function h i (n), to the error function {tilde over (e)}(n), and to the respective input signal M i (n).

10. The system of claim 1 , further comprising a second microphone interface, coupled to the processor, for receiving the second input signal from the second microphone, wherein the second microphone interface is further for providing a sound signal to a speaker that is being used as the second microphone.

11. The system of claim 1 , further comprising a bone conduction microphone interface, coupled to the processor, for receiving the third input signal from the third microphone, wherein the bone conduction microphone interface is further for providing a bone conductible sound signal to a bone conduction speaker that is being used as the bone conduction microphone.

12. The system of claim 1 , wherein the processor is further configured to process sound signals that are detected by multiple bone conduction microphones.

13. The system of claim 1 , wherein the processor is comprised in a mobile communication device, which further comprises the first microphone.

14. The system of claim 1 , further comprising the first microphone, which is configured to transduce an air-carried sound signal, for providing the first input signal.

15. The system of claim 1 , further comprising third microphone, that is configured to transduce a bone-carried sound signal from a bone of a user for providing the third input signal.

16. The system of claim 1 , wherein the processor is further configured to determine an ambient-noise estimation signal, wherein the system further comprises an interface for providing to the user an audio signal that is processed in response to the ambient-noise estimation signal for reducing ambient noise interferences to the user.

17. The system of claim 16 , wherein the processor is further configured to process the audio signal that is provided to the user via bone-conduction speakers in response to the ambient-noise estimation signal and in response to at least one bone-conductivity related parameter.

18. The system of claim 16 , wherein the processor is further configured to update an adaptive noise reduction filter W 1 ( z ), that is used by the processor for processing the audio signal that is provided to the user, in response to the second input signal, wherein the adaptive noise reduction filter W 1 ( z ) corresponds to an estimated audial transformation of sound in an ear canal of the user.

19. The system of claim 16 , wherein the processor is further configured to process an audio signal in response to the ambient-noise estimation signal for reducing ambient noise interferences to the user, wherein the processing of the audio signal is further responsive to a cancellation-level selected by a user of the system.

20. A method for processing sound, the method comprising:

processing a first input signal that is detected by a first microphone at a detection moment, a second input signal that is detected by a second microphone at the detection moment, and a third input signal that is detected by a bone-conduction microphone at the detection moment, to generate a corrected signal that is responsive to the first, second, and third input signals; and providing the corrected signal to an external system; wherein the processing comprises determining the corrected signal Ŝ(n) for the detection moment n, by a sum of convolutions Ŝ(n)=h 1 (n)*M 1 (n)+h 2 (n)*M 2 (n)+h 3 (n)*M 3 (n), wherein M 1 (n) represents the first input signal at the detection moment, M 2 (n) represents the second input signal at the detection moment, M 3 (n) represents the third input signal at the detection moment, and h 1 (n), h 2 (n), and h 3 (n) are calibration functions.

21. The method of claim 20 , wherein the processing is responsive to the second input signal which is detected by the second microphone which is placed at least partly within an ear of a user.

22. The method of claim 21 , wherein the processing is responsive to the second input signal which is transduced by the second microphone from a sound signal which was modified within the ear canal, so that lower frequencies of the sound signal were amplified within the ear canal.

23. The method of claim 21 , wherein the processing is responsive to the second input signal which is detected by the second microphone that is comprised in an ear plug that blocks the ear canal to ambient sound.

24. The method of claim 20 , wherein the processing is preceded by updating at least one calibration function in response to processing of input signals at a past moment that precedes the detection moment.

25. The method of claim 24 , wherein the updating is selectively carried out for a past moment in which a speaking of a user is detected.

26. The method of claim 25 , further comprising detecting a speaking of a user in the past moment by analyzing a speaking spectrum of at least one input signal.

27. The method of claim 24 , wherein the updating is responsive to an error function {tilde over (e)}(n) the value of which for the detection moment n is determined by {tilde over (e)}(n)≈{circumflex over (γ)}(n)*{tilde over (s)}(n)−M 3 (n) where {tilde over (s)}(n) is a sum of H 1 (z), H 2 (z), and H 3 (z), wherein H i (z) is the Z-transform of the corresponding calibration function h i (n).

28. The method of claim 24 , wherein the updating of a calibration function h i (n) is responsive to a partial derivative of a mean square error function J with respect to the calibration function h i (n), to the error function {tilde over (e)}(n), and to the respective input signal M i (n).

29. The method of claim 20 , further comprising providing a sound signal to a speaker that is being used as the second microphone.

30. The method of claim 20 , further comprising providing a bone conductible sound signal to a bone conduction speaker that is being used as the bone conduction microphone.

31. The method of claim 20 , wherein the processing comprises processing Sound signals that are detected by multiple bone conduction microphones.

32. The method of claim 20 , wherein the processing is carried out by a processor which is comprised in a mobile communication device, which further comprises the first microphone.

33. The method of claim 20 , wherein the processing further comprises determining an ambient-noise estimation signal, and processing an audio signal that is provided to the user is response to the ambient-noise estimation signal, for reducing ambient noise interferences to the user.

34. The method of claim 33 , further comprising processing the audio signal that is provided to the user via bone-conduction speakers in response to the ambient-noise estimation signal and in response to at least one bone-conductivity related parameter.

35. The method of claim 33 , wherein the processing of the audio signal that is provided to the user for reducing ambient noise interferences comprises updating an adaptive noise reduction filter W 1 ( z ) that corresponds to an estimated audial transformation of sound in an ear canal of the user in response to the second input signal.

36. The method of claim 33 , wherein processing of the audio signal that is provided to the user for reducing ambient noise interferences is further responsive to a cancellation-level selected by a user of the system.

Assignments (2)
ASSIGNMENT OF ASSIGNOR'S INTEREST Recorded Aug 29, 2023
From: BONETONE COMMUNICATIONS (ISRAEL) LTD
To: DSP GROUP LTD.
Reel/Frame 064743/0194 →
ASSIGNMENT OF ASSIGNOR'S INTEREST Recorded Oct 14, 2013
From: YEHUDAI, URI; HEIMAN, ARIE
To: BONE TONE COMMUNICATIONS LTD.
Reel/Frame 031396/0696 →
Continuity (2)
Provisional Application 61055176 · May 22, 2008
Related Publication 20110135106A1 · Jun 9, 2011