Abstract: In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

Abstract: Near-end equipment for a communication channel with far-end equipment. The near-end equipment includes at least one loudspeaker, at least two microphones, a beamformer, and an echo canceller. The communication channel may be in one of a number of communication states including Near-End Only state, Far-End Only state, and Double-Talk state. In one embodiment, when the echo canceller determines that the communication channel is in either the Far-End Only state or the Double-Talk state, the beamformer is configured to generate a nearfield beampattern signal that directs a null towards a loudspeaker. When the echo canceller detects the Near-End Only state, the beamformer is configured to generate a farfield beampattern signal that optimizes reception of acoustic signals from the near-end audio source. Using different beamformer processing for different communication states allows echo cancellation processing to be more successful at reducing echo in the signal transmitted to the far-end equipment.

Abstract: In one embodiment, an audio system has a microphone array and a signal processing subsystem that processes audio signals generated by the microphone array to produce an output beampattern. The microphone array has (i) a plurality microphones arranged in a circular portion and (ii) a center microphone. The signal processing subsystem has (1) a decomposer that spatially decomposes the microphone audio signals to generate a plurality of eigenbeams and (2) a beamformer that generates the output beampattern as a weighted sum of the eigenbeams. By adding the center microphone, the audio system is able to provide some degree of control over the beamforming in the vertical direction as well as provide reduction of modal aliasin.

Abstract: A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid polyhedron that approximates a sphere. The number and location of the audio sensors on the polyhedron are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene.

Abstract: A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene. Alternative embodiments include using shapes other than spheres, using acoustically soft spheres and/or positioning audio sensors in two or more concentric patterns.

Abstract: In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

Abstract: A microphone array, having a three-dimensional (3D) shape, has a plurality of microphone devices mounted onto (at least one) flexible printed circuit board (PCB), which is bent to achieve the 3D dimensional shape. Output signals from the microphone devices can be combined (e.g., by weighted or unweighted summation or differencing) to form sub-element output signals and/or element output signals, and ultimately a single array output signal for the microphone array. The PCB may be uniformly flexible or may have rigid sections interconnected by flexible portions. Possible 3D shapes include (without limitation) cylinders, spirals, serpentines, and polyhedrons, each formed from a single flexible PCB. Alternatively, the microphone array may be an assembly of multiple, interconnecting sub-arrays, each having two or more rigid portions separated by one or more flexible portions, where each sub-array has at least one cut-out portion for receiving a rigid portion of another sub-array.

Abstract: Near-end equipment for a communication channel with far-end equipment. The near-end equipment includes at least one loudspeaker, at least two microphones, a beamformer, and an echo canceller. The communication channel may be in one of a number of communication states including Near-End Only state, Far-End Only state, and Double-Talk state. In one embodiment, when the echo canceller determines that the communication channel is in either the Far-End Only state or the Double-Talk state, the beamformer is configured to generate a nearfield beampattern signal that directs a null towards a loudspeaker. When the echo canceller detects the Near-End Only state, the beamformer is configured to generate a farfield beampattern signal that optimizes reception of acoustic signals from the near-end audio source. Using different beamformer processing for different communication states allows echo cancellation processing to be more successful at reducing echo in the signal transmitted to the far-end equipment.

Abstract: An audio system generates position-independent auditory scenes using harmonic expansions based on the audio signals generated by a microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of a sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeam outputs. Compensation data corresponding to at least one of the estimated distance and the estimated orientation of the sound source relative to the array are generated from eigenbeam outputs and used to generate an auditory scene. Compensation based on estimated orientation involves steering a beam formed from the eigenbeam outputs in the estimated direction of the sound source to increase direction independence, while compensation based on estimated distance involves frequency compensation of the steered beam to increase distance independence.

Abstract: Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

Abstract: In one embodiment, an audio system has a microphone array and a signal processing subsystem that processes audio signals generated by the microphone array to produce an output beampattem. The microphone array has (i) a plurality microphones arranged in a circular portion and (ii) a center microphone. The signal processing subsystem has (1) a decomposer that spatially decomposes the microphone audio signals to generate a plurality of eigenbeams and (2) a heamformer that generates the output beampattern as a weighted sum of the eigenbeams. By adding the center microphone, the audio system is able to provide some degree of control over the beamforming in the vertical direction as well as provide reduction of modal aliasin.

Abstract: A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene. Alternative embodiments include using shapes other than spheres, using acoustically soft spheres and/or positioning audio sensors in two or more concentric patterns.

Abstract: A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene. Alternative embodiments include using shapes other than spheres, using acoustically soft spheres and/or positioning audio sensors in two or more concentric patterns.

Abstract: In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

Abstract: Spatial noise suppression for audio signals involves generating a ratio of powers of difference and sum signals of audio signals from two microphones and then performing noise suppression processing, e.g., on the sum signal where the suppression is limited based on the power ratio. In certain embodiments, at least one of the signal powers is filtered (e.g., the sum signal power is equalized) prior to generating the power ratio. In a subband implementation, sum and difference signal powers and corresponding the power ratio are generated for different audio signal subbands, and the noise suppression processing is performed independently for each different subband based on the corresponding subband power ratio, where the amount of suppression is derived independently for each subband from the corresponding subband power ratio. In an adaptive filtering implementation, at least one of the audio signals can be adaptively filtered to allow for array self-calibration and modal-angle variability.

Abstract: An audio system generates position-independent auditory scenes using harmonic expansions based on the audio signals generated by a microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of a sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeam outputs. Compensation data corresponding to at least one of the estimated distance and the estimated orientation of the sound source relative to the array are generated from eigenbeam outputs and used to generate an auditory scene. Compensation based on estimated orientation involves steering a beam formed from the eigenbeam outputs in the estimated direction of the sound source to increase direction independence, while compensation based on estimated distance involves frequency compensation of the steered beam to increase distance independence.

Abstract: Two or more microphones receive acoustic signals and generate audio signals that are processed to determine what portion of the audio signals result from (i) incoherence between the audio signals and/or (ii) audio-signal sources having propagation speeds different from the acoustic signals. The audio signals are filtered to reduce that portion of one or more of the audio signals. The present invention can be used to reduce turbulent wind-noise resulting from wind or other airjets blowing across the microphones. Time-dependent phase and amplitude differences between the microphones can be compensated for based on measurements made in parallel with routine audio system processing.

Abstract: Two or more microphones receive acoustic signals and generate audio signals that are processed to determine what portion of the audio signals result from (i) incoherence between the audio signals and/or (ii) audio-signal sources having propagation speeds different from the acoustic signals. The audio signals are filtered to reduce that portion of one or more of the audio signals. The present invention can be used to reduce turbulent wind-noise resulting from wind or other airjets blowing across the microphones. Time-dependent phase and amplitude differences between the microphones can be compensated for based on measurements made in parallel with routine audio system processing.

Abstract: A microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array. In one embodiment, a plurality of audio sensors are mounted on the surface of an acoustically rigid sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene. Alternative embodiments include using shapes other than spheres, using acoustically soft spheres and/or positioning audio sensors in two or more concentric patterns.