Echo cancellation in telephones with multiple microphones

• US 20060147063A1
• Filed: 09/30/2005
• Published: 07/06/2006
• Est. Priority Date: 12/22/2004
• Status: Abandoned Application

• Current Assignee: Avago Technologies General IP PTE Limited (Broadcom, Inc.)
• Original Assignee: Broadcom Corporation (Broadcom, Inc.)
• Inventors: Chen, Juin-Hwey
• Application Number:
  US11/239,351
• Publication Number:
  US 20060147063A1
• Time in Patent Office: Days
• US Class Current: 381/119
• CPC Class Codes: H04M 1/03 Constructional features of telephone transmitters or receivers, e.g. telephone hand-sets
• CPC Class Codes: H04M 1/6008 in the transmitter circuit
• CPC Class Codes: H04M 1/6041 Portable telephones adapted for handsfree use
• CPC Class Codes: H04M 9/082 using echo cancellers echo cancellers per se H04B3/23

Abstract

The present invention is directed to a telephone equipped with multiple microphones that provides improved performance during operation of the telephone in a speaker-phone mode. For example, the multiple microphones can be used to improve voice activity detection, which in turn, can improve echo cancellation. In addition, the multiple microphones can be configured as an adaptive microphone array and used to reduce the effects of (i) room reverberation, when a near-end user is speaking, and/or (ii) acoustic echo, when a far-end user is speaking.

Claims

1. A telephone that provides improved echo cancellation performance, comprising:

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

a loudspeaker that converts the far-end audio signal into sound waves;

a first microphone that picks up the sound waves and outputs a first audio signal, the first audio signal including a first voice component associated with the voice of a near-end user and a second voice component associated with the voice of the far-end user;

a second microphone that outputs a second audio signal;

a voice activity detector (VAD) that processes the first audio signal, the second audio signal and the far-end audio signal to generate output relating to at least one of (i) time intervals in which the first voice component is present in the first audio signal and (ii) time intervals in which the second voice component is present in the first audio signal; and

an echo canceller that cancels the second voice component included in the first audio signal based on the output from the VAD, thereby producing a third audio signal.

2. The telephone of claim 1, wherein the VAD processes the first audio signal, the second audio signal and the far-end audio signal to generate output relating to time intervals in which the first voice component is present in the first audio signal, wherein the processing of the first and second audio signals is based in part on detecting a power ratio between the first and second audio signals.

3. The telephone of claim 1, wherein the echo canceller comprises:

an adaptive filter that adaptively filters the far-end audio signal based on the output from the VAD to produce an adaptively filtered signal; and

a subtraction unit that subtracts the adaptively filtered signal from the first audio signal, thereby producing the third audio signal.

4. The telephone of claim 1, wherein the echo canceller comprises:

an adaptive filter that adaptively filters the second audio signal based on the output from the VAD to produce an adaptively filtered signal; and

a subtraction unit that subtracts the adaptively filtered signal from the first audio signal, thereby producing the third audio signal.

5. The telephone of claim 1, wherein the echo canceller comprises:

an adaptive filter that (i) adaptively filters the far-end audio signal based on the output from the VAD to produce a first adaptively filtered signal, (ii) adaptively filters the second audio signal based on the output from the VAD to produce a second adaptively filtered signal, and (iii) combines the first and second adaptively filtered signals to produce a combined signal; and

a subtraction unit that subtracts the combined signal from the first audio signal, thereby producing the third audio signal.

6. A telephone that provides improved performance, comprising:

a microphone array, each microphone in the array configured to receive sound waves and to generate an audio signal corresponding thereto; and

a digital signal processor (DSP) that receives the audio signals from the microphone array and is configured to detect a direction of arrival (DOA) of at least one of (i) sound waves corresponding to a reverberation of a voice of a near-end user and (ii) sound waves corresponding to a voice signal of a far-end user, wherein the DSP adaptively combines the audio signals from the microphone array based on the DOA to produce a first audio output signal.

7. The telephone of claim 6, wherein the DSP is further configured to adaptively combine the audio signals from the microphone array based on the DOA of sound waves corresponding to the reverberation of the voice of the near-end user to effectively steer a minimum sensitivity angle of the microphone array, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is below a threshold.

8. The telephone of claim 7,

wherein the DSP effectively steers the minimum sensitivity angle of the microphone array so that a source of the reverberation of the voice of the near-end user is within the minimum sensitivity angle.

9. The telephone of claim 6, wherein the DSP is further configured to adaptively combine the audio signals based on the DOA of sound waves corresponding to the voice signal of the far-end user to effectively steer a minimum sensitivity angle of the microphone array, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is below a threshold.

10. The telephone of claim 9,

wherein the DSP effectively steers the minimum sensitivity angle of the microphone array so that a source of the sound waves corresponding to the far-end user'"'"'s voice is within the minimum sensitivity angle.

11. The telephone of claim 6,

wherein the DSP is further configured to detect a DOA of sound waves corresponding to a voice of a near-end user;

and

wherein the DSP is further configured to adaptively combine the audio signals from the microphone array based on the DOA of sound waves corresponding to the voice of the near-end user to effectively steer a maximum sensitivity angle of the microphone array, wherein the maximum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is above a threshold.

12. The telephone of claim 11,

wherein the DSP effectively steers the maximum sensitivity angle of the microphone array so that the near-end user'"'"'s mouth is within the maximum sensitivity angle.

13. The telephone of claim 6,

wherein the DSP is further configured to detect a DOA of sound waves corresponding to a voice of a near-end user;

and

wherein the DSP is further configured (i) to adaptively combine the audio signals based on the DOA of sound waves corresponding to the voice signal of the far-end user to effectively steer a minimum sensitivity angle of the microphone array so that a source of the sound waves corresponding to the far-end user'"'"'s voice is within the minimum sensitivity angle when the far-end user is talking, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is below a threshold, and (ii) to adaptively combine the audio signals based on the DOA of sound waves corresponding to the voice of the near-end user to effectively steer a maximum sensitivity angle of the microphone array so that the near-end user'"'"'s mouth is within the maximum sensitivity angle when the near-end user is talking, wherein the maximum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is above a threshold

14. A method for processing audio signals in a telephone having at least two microphones, comprising:

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

converting the far-end audio signal into sound waves;

picking up the sound waves and outputting a first audio signal from the first microphone, the first audio signal including a first voice component associated with the voice of a near-end user and a second voice component associated with the voice of the far-end user;

outputting a second audio signal from a second microphone;

processing the first audio signal, the second audio signal and the far-end audio signal in a voice activity detector (VAD) to generate output relating to at least one of (i) time intervals in which the first voice component is present in the first audio signal and (ii) time intervals in which the second voice component is present in the first audio signal; and

canceling the second voice component included in the first audio signal based on the output of the VAD, thereby producing a third audio signal.

15. The method of claim 14, wherein processing the first audio signal and the second audio signal comprises:

detecting a power ratio between the first and second audio signals.

16. The method of claim 14, wherein canceling the second voice component included in the first audio signal based on the output of the VAD comprises:

adaptively filtering the far-end audio signal based on the output from the VAD to produce an adaptively filtered signal; and

subtracting the adaptively filtered signal from the first audio signal, thereby producing the third audio signal.

17. The method of claim 14, wherein canceling the second voice component included in the first audio signal based on the output of the VAD comprises:

adaptively filtering the second audio signal based on the output from the VAD to produce an adaptively filtered signal; and

subtracting the adaptively filtered signal from the first audio signal, thereby producing the third audio signal.

18. The method of claim 14, wherein canceling the second voice component included in the first audio signal based on the output of the VAD comprises:

adaptively filtering the far-end audio signal based on the output from the VAD to produce a first adaptively filtered signal;

adaptively filtering the second audio signal based on the output from the VAD to produce a second adaptively filtered signal;

combining the first and second adaptively filtered signals to produce a combined signal; and

subtracting the combined signal from the first audio signal, thereby producing the third audio signal.

19. A method for processing audio signals in a telephone having a microphone array, comprising:

(a) outputting an audio signal from each microphone in the microphone array based on sound waves incident thereon;

(b) detecting a direction of arrival (DOA) of at least one of (i) sound waves corresponding to a reverberation of a near-end user'"'"'s voice and (ii) sound waves corresponding to a far-end user'"'"'s voice; and

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

20. The method of claim 19, wherein step (c) further comprises:

adaptively combining the audio signals based on the DOA of sound waves corresponding to the reverberation of the near-end user'"'"'s voice to effectively steer a minimum sensitivity angle of the microphone array, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is below a threshold.

21. The method of claim 20, further comprising:

adaptively combining the audio signals based on the DOA of sound waves corresponding to the reverberation of the near-end user'"'"'s voice to effectively steer the minimum sensitivity angle of the microphone array so that a source of the reverberation of the near-end user'"'"'s voice is within the minimum sensitivity angle.

22. The method of claim 19, wherein step (c) further comprises:

adaptively combining the audio signals based on the DOA of sound waves corresponding to the far-end user'"'"'s voice to effectively steer a minimum sensitivity angle of the microphone array, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is below a threshold.

23. The method of claim 22, further comprising:

adaptively combining the audio signals based on the DOA of sound waves corresponding to the far-end user'"'"'s voice to effectively steer the minimum sensitivity angle of the microphone array so that a source of the sound waves corresponding to the far-end user'"'"'s voice is within the minimum sensitivity angle.

24. The method of claim 19, wherein steps (b) and (c) further comprise:

(b1) detecting a DOA of sound waves corresponding to a voice of a near-end user; and

(c1) adaptively combining the audio signals from the microphone array based on the DOA of sound waves corresponding to the voice of the near-end user to effectively steer a maximum sensitivity angle of the microphone array, wherein the maximum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is above a threshold.

25. The method of claim 24, further comprising:

effectively steering the maximum sensitivity angle of the microphone array so that the near-end user'"'"'s mouth is within the maximum sensitivity angle.

26. The method of claim 19, wherein steps (b) and (c) further comprise:

(b1) detecting a DOA of sound waves corresponding to a voice of a near-end user;

(c1) adaptively combining the audio signals based on the DOA of sound waves corresponding to the voice signal of the far-end user to effectively steer a minimum sensitivity angle of the microphone array so that a source of the sound waves corresponding to the far-end user'"'"'s voice is within the minimum sensitivity angle when the far-end user is talking, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is below a threshold; and

(c2) adaptively combining the audio signals based on the DOA of sound waves corresponding to the voice of the near-end user to effectively steer a maximum sensitivity angle of the microphone array so that the near-end user'"'"'s mouth is within the maximum sensitivity angle when the near-end user is talking, wherein the maximum sensitivity angle is defined as an angle within which a sensitivity of the microphone array is above a threshold.