Abstract: In one embodiment, a directional microphone array having (at least) two microphones mounted on opposite sides of a device generates forward and backward base signals from two (e.g., omnidirectional) microphone signals using diffraction filters and equalization filters. Each diffraction filter implements a (possibly different) transfer function representing the response of an audio signal traveling from a corresponding microphone around the device to the other microphone. A scale factor is applied to, for example, the backward base signal, and the resulting scaled backward base signal is combined with (e.g., subtracted from) the forward base signal to generate a first-order differential audio signal. After low-pass filtering, spatial noise suppression can be applied to the first-order differential audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order differential audio signals.

Abstract: Accessories for a telephone include at least one earphone and at least one microphone array having multiple microphones used to generate outgoing audio signals for (i) processing by a signal processor and (ii) transmission by the telephone. In one embodiment, two earphones are connected by two corresponding wires, and two microphone arrays, respectively connected to the two wires, are mechanically and electronically configurable in a plurality of use modes to generate outgoing audio signals for processing by the signal processor. The use modes include one or more and possibly all of a single-sided mode, a two-sided mode, an enhanced directivity mode, a stereo recording mode, a multichannel recording mode, a conference mode, and a two-dimensional-array mode, where one of the use modes is automatically detected by the signal processor based on the audio signals generated by the two microphone arrays.

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 first microphone set of two or more microphones located on a first ellipse, (ii) a second microphone set of two or more microphones located on a second ellipse within the first ellipse, and (iii) a third microphone set of one or more microphones located within the second ellipse, where the microphones in the first, second, and third microphone sets are effectively all in one plane. 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.

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, 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: 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: 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: A microphone device is provided which comprises a main microphone (MM), at least one control microphone (CM) and a digital signal processing unit (DSP) coupled to the main microphone (MM) and the at least one control microphone (CM). The digital signal processing unit (DSP) receives the output of the main microphone (MM) and the output of the at least one control microphone (CM). Based on the output signals, the digital signal processing unit (DSP) is adapted to perform a noise suppression of pop noise in the output signal of the main microphone (MM).

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 device is provided which comprises a main microphone (MM), at least one control microphone (CM) and a digital signal processing unit (DSP) coupled to the main microphone (MM) and the at least one control microphone (CM). The digital signal processing unit (DSP) receives the output of the main microphone (MM) and the output of the at least one control microphone (CM). Based on the output signals, the digital signal processing unit (DSP) is adapted to perform a noise suppression of pop noise in the output signal of the main microphone (MM).

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: 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: 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.