Abstract: A mobile communication device and a method of mitigating wind noise in a speech signal generated at a microphone of the mobile communication device. A primary microphone of the mobile communication device receives speech and wind noise and generates a first speech signal based on the speech and the wind noise that is received. A secondary microphone of the mobile communication device receives attenuated speech and generates a second speech signal based on the attenuated speech that is received. The secondary microphone is shielded from the wind noise when the primary microphone receives speech and the wind noise. A processor of the mobile communication device processes the first speech signal to replace a low frequency portion of the first speech signal with the second speech signal in order to mitigate the wind noise in the first speech signal.

Abstract: A mobile communication device and a method of mitigating wind noise in a speech signal generated at a microphone of the mobile communication device. A primary microphone of the mobile communication device receives speech and wind noise and generates a first speech signal based on the speech and the wind noise that is received. A secondary microphone of the mobile communication device receives attenuated speech and generates a second speech signal based on the attenuated speech that is received. The secondary microphone is shielded from the wind noise when the primary microphone receives speech and the wind noise. A processer of the mobile communication device processes the first speech signal to replace a low frequency portion of the first speech signal with the second speech signal in order to mitigate the wind noise in the first speech signal.

Abstract: Audio input may be received at one or more microphones of a mobile device. Based on the audio input, a change in an acoustic environment of the device, such as a change in a direction of arrival of the audio input or a degradation in a quality of acoustic echo cancellation being performed upon the audio input, may be detected. A determination may be made that the detected change coincides with a non-acoustic physical event detected using an auxiliary sensor at the mobile device. The event may for example be device motion, a new proximate object, a change in a proximity of an object, a new heat source or a change in a heat level from a known heat source. Based on the determining, a signal processor, possibly comprising an audio beamformer or echo canceller, may be recalibrated, e.g. the audio beamformer or echo canceller may be caused to reconverge.

Abstract: A telephonic device having a speakerphone function has a loudspeaker and a plurality of microphones. The plurality of microphones are coupled to a plurality of beamformers which produce a first beamform having a spatial null in the direction of the loudspeaker and a second beamform having a spatial null in the direction of near-end signals. The two beamforms are then processed to facilitate echo reduction. By comparing the beamform powers, a state of double-talk can be determined and the determination can be used to enhance echo reduction associated with speakerphone functionality.

Abstract: A telephonic device having a speakerphone function has a loudspeaker and a plurality of microphones. The plurality of microphones are coupled to a plurality of beamformers which produce a first beamform having a spatial null in the direction of the loudspeaker and a second beamform having a spatial null in the direction of near-end signals. The two beamforms are then processed to facilitate echo reduction. By comparing the beamform powers, a state of double-talk can be determined and the determination can be used to enhance echo reduction associated with speakerphone functionality.

Abstract: Audio input may be received at one or more microphones of a mobile device. Based on the audio input, a change in an acoustic environment of the device, such as a change in a direction of arrival of the audio input or a degradation in a quality of acoustic echo cancellation being performed upon the audio input, may be detected. A determination may be made that the detected change coincides with a non-acoustic physical event detected using an auxiliary sensor at the mobile device. The event may for example be device motion, a new proximate object, a change in a proximity of an object, a new heat source or a change in a heat level from a known heat source. Based on the determining, a signal processor, possibly comprising an audio beamformer or echo canceller, may be recalibrated, e.g. the audio beamformer or echo canceller may be caused to reconverge.

Abstract: A method and apparatus is set forth for accelerating the total acoustic echo cancellation convergence time in a microphone array full-duplex conferencing system with beamformer. The invention is based on sequentially switching the beamformer to untrained sectors during periods of far-end speech activity and performing real-time cancellation of far-end echo signals from the near-end signals to consecutively adapt and store filter coefficients for the echo canceller corresponding to each sector.

Abstract: A method is set forth for reducing the total acoustic echo cancellation convergence time for all look directions in a microphone array based full-duplex system. The method is based on capturing the loudspeaker signal due to the first far-end speech bursts when the conferencing system is first used, as well as the corresponding loudspeaker feedback signals in the individual microphones. The captured signals are then used for consecutive adaptation of the acoustic echo canceller on all echo paths corresponding to all look directions of the beamformer, thereby training the AEC. This training process can be executed concurrently with normal phone operation, for example, as a background process that utilizes available processing cycles.

Abstract: A method to reduce acoustic coupling in an audio conferencing system having a loudspeaker and a plurality of microphones, comprising detecting the presence of one of either a source of near-end signal or a source of far-end signal source relative to the audio conferencing system, and in the event of detecting a source of near-end signal then processing the near-end signal with high directivity towards the near-end source, and in the event of detecting a source of far-end signal or double-talk then processing the far-end signal with high coupling characteristics between the loudspeaker and microphones.

Abstract: A method of designing a beamformer, and a beamformer made in accordance with this method, characterized by a uniform speakerphone response condition, resulting in optimal beamforming directivity under a uniform coupling constraint. According to the present invention, a finite number of individual beamformers are constrained to have the same response to a loudspeaker signal (as well as the same gain in their respective look directions) without specifying the exact value of their response to this signal. This results in beamformer weights that are optimal in the minimum variance sense and satisfy the uniform coupling constraint. The minimum variance condition combines all beamformer weights at once, and the uniform coupling constraint is expressed as a finite number of linear constraints on the weights of the individual beamformers, without specifying an arbitrary, a priori value for the actual value of the uniform response.

Abstract: An apparatus for locating a talker, comprising a microphone array for receiving multiple audio signals, wherein the microphone array is characterized by a predetermined beampattern, a spectral conditioner for filtering the audio signals to optimize the beampattern for talker localization, a localization estimator for calculating a localization estimate based on the filtered audio signals, an activity detector for detecting periods of speech activity, and decision logic for verifying the localization estimate during the periods of speech activity.

Abstract: An apparatus for locating a talker, comprising a microphone array for receiving multiple audio signals, wherein the microphone array is characterized by a predetermined beampattern, a spectral conditioner for filtering the audio signals to optimize the beampattern for talker localization, a localization estimator for calculating a localization estimate based on the filtered audio signals, an activity detector for detecting periods of speech activity, and decision logic for verifying the localization estimate during the periods of speech activity.

Abstract: A method of improving convergence of an echo canceller in a full duplex speakerphone, wherein the echo canceller includes LEC (Line Echo Canceller) and AEC (Acoustic Echo Canceller) portions, comprising the steps of capturing coefficients during operation, storing the coefficients, and utilizing the stored coefficients as default values during start-up of a subsequent call. According to the method of the present invention, the amount of echo cancellation necessary to provoke a save of the coefficients decreases with time when it is not achieved by the system despite the presence of a reference signal (i.e. speakerphone signal).

Abstract: A method is set forth for reducing the total acoustic echo cancellation convergence time for all look directions in a microphone array based full-duplex system. The method is based on capturing the loudspeaker signal due to the first far-end speech bursts when the conferencing system is first used, as well as the corresponding loudspeaker feedback signals in the individual microphones. The captured signals are then used for consecutive adaptation of the acoustic echo canceller on all echo paths corresponding to all look directions of the beamformer, thereby training the AEC. This training process can be executed concurrently with normal phone operation, for example, as a background process that utilizes available processing cycles.

Abstract: A method of locating a talker in a reverberant environment comprises receiving multiple audio signals from a microphone array that include direct path audio signal and reverberation signal components. The direct path audio signal components of the multiple audio signals are detected and are used to weight the multiple audio signals. A position estimate based on the weighted audio signals is then calculated. Periods of speech activity are detected and a final position estimate is generated during the periods of speech activity.

Abstract: A method and apparatus is set forth for accelerating the total acoustic echo cancellation convergence time in a microphone array full-duplex conferencing system with beamformer. The invention is based on sequentially switching the beamformer to untrained sectors during periods of far-end speech activity and performing real-time cancellation of far-end echo signals from the near-end signals to consecutively adapt and store filter coefficients for the echo canceller corresponding to each sector.

Abstract: Apparatus for controlling noise characteristic estimation in a conferencing system, comprising a noise characteristic estimator for estimating a noise characteristic of a signal of interest transmitted in a first direction through the conferencing system, and a first voice activity detector for detecting audio signal activity in a signal transmitted through the conferencing system in a direction opposite to the signal of interest and in response disabling the noise characteristic estimator.

Abstract: A method is set forth of controlling an acoustic echo canceller at the output of a beamformer in an audio conferencing device. Information is saved to, and retrieved from, memory that characterizes each of a finite number of look directions, or regions of focus, covering the entire spatial span of the conferencing device. Each time a change occurs from a first look direction to a second look direction, information relating to the workspace captured by the acoustic echo canceller is saved for the first look direction, and previously saved information for the second look direction is retrieved from memory. The acoustic echo cancellation then takes place for the second look direction with the retrieved information.

Abstract: A method of designing a beamformer, and a beamformer made in accordance with this method, characterized by a uniform speakerphone response condition, resulting in optimal beamforming directivity under a uniform coupling constraint. According to the present invention, a finite number of individual beamformers are constrained to have the same response to a loudspeaker signal (as well as the same gain in their respective look directions) without specifying the exact value of their response to this signal. This results in beamformer weights that are optimal in the minimum variance sense and satisfy the uniform coupling constraint. The minimum variance condition combines all beamformer weights at once, and the uniform coupling constraint is expressed as a finite number of linear constraints on the weights of the individual beamformers, without specifying an arbitrary, a priori value for the actual value of the uniform response.

Abstract: A method to reduce acoustic coupling in an audio conferencing system having a loudspeaker and a plurality of microphones, comprising detecting the presence of one of either a source of near-end signal or a source of far-end signal source relative to the audio conferencing system, and in the event of detecting a source of near-end signal then processing the near-end signal with high directivity towards the near-end source, and in the event of detecting a source of far-end signal or double-talk then processing the far-end signal with high coupling characteristics between the loudspeaker and microphones.