Wireless telephone with adaptive microphone array

• US 7,983,720 B2
• Filed: 05/24/2005
• Issued: 07/19/2011
• Est. Priority Date: 12/22/2004
• 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): Appiah; Charles N
• Assistant Examiner(s): STEPHEN, EMEM O
• Application Number:
  US11/135,491
• Publication Number:
  US 20060133622A1
• Time in Patent Office: 2,247 Days
• Field of Search: 4555751-5754, 455/575.7, 455/550, 455/90.3, 455/347, 381/92, 381/122
• US Class Current: 455/575.1
• 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

A wireless telephone having an array of microphones and a digital signal processor (DSP) and a method of processing audio signals from a wireless telephone having an array of microphones and a DSP. The wireless telephone includes an array of microphones and a DSP. Each microphone in the array is configured to receive sound waves emanating from the surrounding environment and to generate an audio signal corresponding thereto. The DSP is coupled to the array of microphones. The DSP is configured to receive the audio signals from the array of microphones, to detect a direction of arrival (DOA) of a sound wave emanating from the mouth of a user based on the audio signals and to adaptively combine the audio signals based on the DOA to produce a first audio output signal.

Claims

1. A wireless telephone, comprising:

a first array of microphones, each microphone in the first array of microphones configured to receive sound waves emanating from the surrounding environment and to generate an audio signal corresponding thereto;

a first digital signal processor (DSP) coupled to the first array of microphones, the first DSP configured to receive the audio signals from the first array of microphones and to detect a direction of arrival (DOA) of a sound wave emanating from a background noise source based on the audio signals and to adaptively combine the audio signals based on the DOA to effectively steer a minimum sensitivity angle of the first microphone array so that the background noise source is within the minimum sensitivity angle so as to produce a first audio output signal, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the first microphone array is below a predetermined threshold;

a second array of microphones, each microphone in the second array of microphones configured to receive sound waves emanating from the surrounding environment and to generate an audio signal corresponding thereto;

a second DSP coupled to the second array of microphones, the second DSP configured to receive the audio signals from the second array of microphones and to detect a DOA of the sound wave emanating from the background noise source based on the audio signals and to adaptively combine the audio signals based on the DOA to produce a second audio output signal; and

a voice activity detector (VAD) that receives the first and second audio output signals and detects time intervals in which a voice component is present in the first audio output signal based on the content of the first audio output signal and the second audio output signal,

wherein the second DSP is further configured to detect a DOA of a sound wave emanating from the mouth of a user and to dynamically switch between one of

(i) adaptively combining the audio signals from the second array of microphones based on the DOA of the sound wave emanating from the mouth of the user to effectively steer a minimum sensitivity angle of the second microphone array so that the mouth of the user is within the minimum sensitivity angle during time intervals in which a voice component is present in the first audio output signal, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the second microphone array is below a predetermined threshold, and

(ii) adaptively combining the audio signals from the second array of microphones based on the DOA of the background noise source to effectively steer a maximum sensitivity angle of the second microphone array so that the background noise source is within the maximum sensitivity angle during time intervals in which a voice component is not present in the first audio output signal, wherein the maximum sensitivity angle is defined as an angle within which a sensitivity of the second microphone array is above a predetermined threshold.

2. A method for processing audio signals in a wireless telephone comprising:

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

(b) detecting a direction of arrival (DOA) of a sound wave emanating from a background noise source based on the audio signals;

(c) adaptively combining the audio signals based on the DOA to effectively steer a minimum sensitivity angle of the first microphone array so that the background noise source is within the minimum sensitivity angle so as to produce a first audio output signal, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the first microphone is below a predetermined threshold;

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

(e) adaptively combining the audio signals from the microphones in the second microphone array based on the DOA to produce a second audio output signal;

(f) detecting time intervals in which a voice component is present in the first audio output signal based on the content of the first audio output signal and the second audio output signal; and

(g) detecting a DOA of a sound wave emanating from the mouth of a user; and

(h) dynamically switching between one of

(1) adaptively combining the audio signals from the second array of microphones based on the DOA of the sound wave emanating from the mouth of the user to effectively steer a minimum sensitivity angle of the second microphone array so that the mouth of the user is within the minimum sensitivity angle during time intervals in which a voice component is present in the first audio output signal, wherein the minimum sensitivity angle is defined as an angle within which a sensitivity of the second microphone array is below a predetermined threshold, and

(2) adaptively combining the audio signals from the second array of microphones based on the DOA of the background noise source to effectively steer a maximum sensitivity angle of the second microphone array so that the background noise source is within the maximum sensitivity angle during time intervals in which a voice component is not present in the first audio output signal, wherein the maximum sensitivity angle is defined as an angle within which a sensitivity of the second microphone array is above a predetermined threshold.