Speech intelligibility in telephones with multiple microphones

• US 8,428,661 B2
• Filed: 10/30/2007
• Issued: 04/23/2013
• Est. Priority Date: 10/30/2007
• Status: Active Grant

• Current Assignee: Avago Technologies International Sales Pte Limited (Broadcom, Inc.)
• Original Assignee: Broadcom Corporation (Broadcom, Inc.)
• Inventors: Chen, Juin-Hwey
• Primary Examiner(s): Gelin, Jean
• Application Number:
  US11/976,995
• Publication Number:
  US 20090111507A1
• Time in Patent Office: 2,002 Days
• Field of Search: 455/570, 455/575.1, 455/575.7, 455/90.3, 455/425, 455/66.1, 455/67.13, 455/63.1, 455/569.1, 455/501, 455/463, 704/226, 704/227, 704/228, 704 941- 949, 704/233, 704/246, 704/275, 381/92, 381/122, 379/392.01
• US Class Current: 455/570
• CPC Class Codes: H04M 1/6008 in the transmitter circuit
• CPC Class Codes: H04R 3/005 for combining the signals of two or more microphones specially adapted for hearing aids H04R25/407

Abstract

The present invention is directed to improved speech intelligibility in telephones with multiple microphones. Such a telephone includes a first microphone, a second microphone, a voice activity detector (VAD), a receiver module, and a signal processor. The first microphone outputs a first audio signal, which comprises a voice component when a near-end user talks and a background noise component. The second microphone outputs a second audio signal. The VAD generates a voice activity signal responsive to a ratio between the first audio signal and the second audio signal. The voice activity signal identifies time intervals in which the voice component of the near-end user is present in the first audio signal. The receiver module receives a third audio signal, which comprises a voice component of a far-end user. The signal processor modifies the third audio signal responsive to the voice activity signal.

Claims

1. A method in a telephone for improving speech intelligibility when the telephone is used in an environment having background noise, the method comprising:

outputting a first audio signal from a first microphone, the first audio signal comprising a voice component when a near-end user talks and a background noise component;

outputting a second audio signal from a second microphone;

monitoring a ratio between the first audio signal and the second audio signal;

generating a voice activity signal responsive to the ratio between the first audio signal and the second audio signal, the voice activity signal identifying time intervals in which the first audio signal includes the voice component of the near-end user;

receiving a third audio signal comprising a voice component of a far-end user; and

modifying the third audio signal only during time intervals when the near-end user is not talking as indicated by the voice activity signal.

2. The method of claim 1, wherein the modifying comprises:

boosting the third audio signal responsive to the voice activity signal.

3. The method of claim 1, wherein the modifying comprises:

comparing a frequency spectrum of the third audio signal to a frequency spectrum of the second audio signal; and

boosting at least one frequency of the third audio signal responsive to the voice activity signal.

4. The method of claim 1, further comprising:

canceling the background noise component of the first audio signal based on the content of the first audio signal and the second audio signal.

5. The method of claim 1, further comprising:

modifying the third audio signal based on the content of the third audio signal and the content of the second audio signal; and

generating a sound wave from a loudspeaker based on the modified third audio signal, the sound wave reducing the background noise of the environment proximate to the loudspeaker.

6. The method of claim 1, wherein the first microphone comprises a microphone array.

7. The method of claim 6, wherein outputting the first audio signal from the microphone array comprises:

(a) outputting an audio signal from each microphone in the microphone array;

(b) detecting a direction of arrival (DOA) of a sound wave emanating from the mouth of the near-end user based on the audio signals; and

(c) adaptively combining the audio signals based on the DOA to produce the first audio signal.

8. The method of claim 7, wherein step (c) comprises:

effectively steering a maximum sensitivity angle of the microphone array so that the mouth of the near-end user is within the maximum sensitivity angle, wherein the maximum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is above a predetermined threshold.

9. The method of claim 1, wherein the second microphone comprises a microphone array.

10. The method of claim 9, wherein outputting the second audio signal from the microphone array comprises:

(a) outputting an audio signal from each microphone in the microphone array;

(b) detecting a direction of arrival (DOA) of a sound wave emanating from the mouth of the near-end user based on the audio signals; and

(c) adaptively combining the audio signals based on the DOA to produce the second audio signal.

11. The method of claim 10, wherein step (c) comprises:

effectively steering a minimum sensitivity angle of the microphone array so that the mouth of the near-end user is within the minimum sensitivity angle, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is below a predetermined threshold.

12. A telephone for improving speech intelligibility when the telephone is used in an environment having background noise, the telephone comprising:

a first microphone that outputs a first audio signal, the first audio signal comprising a voice component when a near-end user talks and a background noise component;

a second microphone that outputs a second audio signal;

a voice activity detector (VAD) that (i) monitors a ratio between the first audio signal and the second audio signal, and (ii) generates a voice activity signal responsive to the ratio between the first audio signal and the second audio signal, wherein the voice activity signal identifies time intervals in which the first audio signal includes the voice component of the near-end user;

a receiver module that receives a third audio signal, the third audio signal comprising a voice component of a far-end user; and

a signal processor that modifies the third audio signal only during time intervals when the near-end user is not talking as indicated by the voice activity signal.

13. The telephone of claim 12, wherein the signal processor comprises:

signal processing logic that boosts the third audio signal responsive to the voice activity signal.

14. The telephone of claim 12, wherein the signal processor comprises:

signal processing logic that (i) compares a frequency spectrum of the third audio signal to a frequency spectrum of the second audio signal, and (ii) boosts at least one frequency of the third audio signal responsive to the voice activity signal.

15. The telephone of claim 12, further comprising:

a noise cancellation module that cancels the background noise component of the first audio signal based on the content of the first audio signal and the second audio signal.

16. The telephone of claim 12, further comprising:

a noise cancellation module that modifies the third audio signal based on the content of the third audio signal and the content of the second audio signal; and

a loudspeaker that generates a sound wave based on the modified third audio signal, the sound wave reducing the background noise of the environment proximate to the loudspeaker.

17. The telephone of claim 12, wherein the first microphone comprises a microphone array.

18. The telephone of claim 17, wherein each microphone of the microphone array outputs an audio signal, and the telephone further comprises:

a digital signal processor (DSP) that (i) detects a direction of arrival (DOA) of a sound wave emanating from the mouth of the near-end user based on the audio signals output by the microphones of the microphone array, and (ii) adaptively combines the audio signals based on the DOA to produce the first audio signal.

19. The telephone of claim 18, wherein the DSP effectively steers a maximum sensitivity angle of the microphone array so that the mouth of the near-end user is within the maximum sensitivity angle, wherein the maximum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is above a predetermined threshold.

20. The telephone of claim 12, wherein the second microphone comprises a microphone array.

21. The telephone of claim 20, wherein each microphone of the microphone array outputs an audio signal, and the telephone further comprises:

a digital signal processor (DSP) that (i) detects a direction of arrival (DOA) of a sound wave emanating from the mouth of the near-end user based on the audio signals output by the microphones of the microphone array, and (ii) adaptively combines the audio signals based on the DOA to produce the second audio signal.

22. The telephone of claim 21, wherein the DSP effectively steers a minimum sensitivity angle of the microphone array so that the mouth of the near-end user is within the minimum sensitivity angle, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is below a predetermined threshold.

23. A method in a telephone for improving speech intelligibility when the telephone is used in an environment having background noise, comprising:

outputting a first audio signal from a first microphone, the first audio signal comprising a voice component when a near-end user talks and a background noise component;

reversing a polarity of the first audio signal to form a second audio signal;

receiving a third audio signal comprising a voice component of a far-end user; and

forming a side-tone based on the second audio signal and the third audio signal.

24. A telephone for improving speech intelligibility when the telephone is used in an environment having background noise, comprising:

a first microphone that outputs a first audio signal, the first audio signal comprising a voice component when a near-end user talks and a background noise component;

a signal processor that reverses a polarity of the first audio signal to form a second audio signal;

a receiver module that receives a third audio signal comprising a voice component of a far-end user; and

a speaker that outputs a side-tone that is formed based on the second audio signal and the third audio signal.