Abstract: According to certain embodiments, a microphone array having a plurality of microphone elements is calibrated by ensonifying the microphone array at a first direction relative to the microphone array with a first acoustic signal to concurrently generate a first set of audio signals from two or more of the microphone elements and processing the first set of audio signals to calibrate the two or more microphone elements. One or more other sets of audio signals can be generated by ensonifying the microphone array with one or more other acoustic signals at one or more other directions relative to the microphone array, where the two or more microphone elements are calibrated using the first set and the one or more other sets of audio signals. The calibration process can be performed outside of an anechoic chamber using one or more acoustic sources located outside or inside the microphone array.

Abstract: According to certain embodiments, a microphone array having a plurality of microphone elements is calibrated by ensonifying the microphone array at a first direction relative to the microphone array with a first acoustic signal to concurrently generate a first set of audio signals from two or more of the microphone elements and processing the first set of audio signals to calibrate the two or more microphone elements. One or more other sets of audio signals can be generated by ensonifying the microphone array with one or more other acoustic signals at one or more other directions relative to the microphone array, where the two or more microphone elements are calibrated using the first set and the one or more other sets of audio signals. The calibration process can be performed outside of an anechoic chamber using one or more acoustic sources located outside or inside the microphone array.

Abstract: In one embodiment, an article of manufacture has microphones mounted at different locations on a non-spheroidal device body and a signal-processing system that processes the microphone signals to generate a B Format audio output having a zeroth-order beampattern signal and three first-order beampattern signals in three orthogonal directions. The signal-processing system generates at least one of the first-order beampattern signals based on effects of the device body on an incoming acoustic signal. The microphone signals used to generate each first-order beampattern signal have an inter-microphone effective distance that is less than a wavelength at a specified high-frequency value (e.g., <4 cm for 8 kHz). In preferred embodiments, the inter-microphone effective distance is less than one-half of that wavelength (e.g., <2 cm for 8 kHz).

Abstract: In one embodiment, an article of manufacture has microphones mounted at different locations on a non-spheroidal device body and a signal-processing system that processes the microphone signals to generate a B Format audio output having a zeroth-order beampattern signal and three first-order beampattern signals in three orthogonal directions. The signal-processing system generates at least one of the first-order beampattern signals based on effects of the device body on an incoming acoustic signal. The microphone signals used to generate each first-order beampattern signal have an inter-microphone effective distance that is less than a wavelength at a specified high-frequency value (e.g., <4 cm for 8 kHz). In preferred embodiments, the inter-microphone effective distance is less than one-half of that wavelength (e.g., <2 cm for 8 kHz).

Abstract: In one embodiment, an article of manufacture has microphones mounted at different locations on a non-spheroidal device body and a signal-processing system that processes the microphone signals to generate a B Format audio output having a zeroth-order beampattern signal and three first-order beampattern signals in three orthogonal directions. The signal-processing system generates at least one of the first-order beampattern signals based on effects of the device body on an incoming acoustic signal. The microphone signals used to generate each first-order beampattern signal have an inter-microphone effective distance that is less than a wavelength at a specified high-frequency value (e.g., <4 cm for 8 kHz). In preferred embodiments, the inter-microphone effective distance is less than one-half of that wavelength (e.g., <2 cm for 8 kHz).

Abstract: In one embodiment, an article of manufacture has microphones mounted at different locations on a non-spheroidal device body and a signal-processing system that processes the microphone signals to generate a B Format audio output having a zeroth-order beampattern signal and three first-order beampattern signals in three orthogonal directions. The signal-processing system generates at least one of the first-order beampattern signals based on effects of the device body on an incoming acoustic signal. The microphone signals used to generate each first-order beampattern signal have an inter-microphone effective distance that is less than a wavelength at a specified high-frequency value (e.g., <4 cm for 8 kHz). In preferred embodiments, the inter-microphone effective distance is less than one-half of that wavelength (e.g., <2 cm for 8 kHz).

Abstract: In certain embodiments, an article of manufacture, such as a cell phone, has a device body with a non-spheroidal shape, such as a parallelepiped, and microphones configured at different locations on the device body. A signal processing system processes the microphone signals to generate a plurality of different output beampatterns in at least two non-parallel directions, wherein, in generating at least one of the output beampatterns, the signal processing system takes into account effects of the device body on the incoming acoustic signal. Four or more microphones can be used to generate B format output beampatterns, such as three dipole beampatterns and an omnidirectional beampattern.

Abstract: In one embodiment, an audio processing system reduces reverberation in an audio signal. A first beamformer generates a first, directional beampattern, and a second beamformer generates a second beampattern. A signal-processing subsystem (i) processes the first and second beampatterns to generate suppression factors corresponding to the reverberation and (ii) applies the suppression factors to one of the first and second beampatterns to reduce the reverberation in the beampattern. In one implementation, the beampatterns are crossed-beam beampatterns, and the signal-processing subsystem generates the suppression factors based on coherence estimates for the beampatterns. In another implementation, the beampatterns are disjoint beampatterns, and the signal-processing subsystem generates the suppression factors based on short-time and long-time envelope estimates for the beampatterns.

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 processing system reduces reverberation in an audio signal. A first beamformer generates a first, directional beampattern, and a second beamformer generates a second beampattern. A signal-processing subsystem (i) processes the first and second beampatterns to generate suppression factors corresponding to the reverberation and (ii) applies the suppression factors to one of the first and second beampatterns to reduce the reverberation in the beampattern. In one implementation, the beampatterns are crossed-beam beampatterns, and the signal-processing subsystem generates the suppression factors based on coherence estimates for the beampatterns. In another implementation, the beampatterns are disjoint beampatterns, and the signal-processing subsystem generates the suppression factors based on short-time and long-time envelope estimates for the beampatterns.

Abstract: In certain embodiments, an article of manufacture, such as a cell phone, has a device body with a non-spheroidal shape, such as a parallelepiped, and microphones configured at different locations on the device body. A signal processing system processes the microphone signals to generate a plurality of different output beampatterns in at least two non-parallel directions, wherein, in generating at least one of the output beampatterns, the signal processing system takes into account effects of the device body on the incoming acoustic signal. Four or more microphones can be used to generate B format output beampatterns, such as three dipole beampatterns and an omnidirectional beampattern.

Abstract: An exemplary audio signal processing system includes a modal decomposer and an adaptive modal beamformer. The modal decomposer generates a plurality of zeroth-order eigenbeams from audio signals from an (e.g., spherical) array of audio sensors. The adaptive modal beamformer (i) steers the zeroth-order eigenbeams to a specified direction, (ii) adaptively generates a plurality of weighting coefficients for the plurality of zeroth-order eigenbeams, where the plurality of weighting coefficients satisfy a constraint of having only non-negative values, (iii) respectively applies the plurality of adaptively generated weighting coefficients to the plurality of steered, zeroth-order eigenbeams to generate a plurality of weighted, steered, zeroth-order eigenbeams, and (iv) combines the plurality of weighted, steered, zeroth-order eigenbeams to generate an output audio signal. Some embodiments have a further constraint that the weighting coefficients sum to a specified value (e.g., one).

Abstract: In one embodiment, 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: 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 system has an output and at least a first transducer with a first dynamic range, a second transducer with a second dynamic range different than the first dynamic range, and coupling system to selectively couple the output of one of the first transducer or the second transducer to the system output, depending on the magnitude of the input sound signal, to produce a system with a dynamic range greater than the dynamic range of either individual transducer. A method of operating a microphone system includes detecting whether a transducer output crosses a threshold, and if so then selectively coupling another transducer's output to the system output. The threshold may change as a function of which transducer is coupled to the system output. The system and methods may also combine the outputs of more than one transducer in a weighted sum during transition from one transducer output to another, as a function of time or as a function of the amplitude of the incident audio signal.

Abstract: An exemplary audio signal processing system includes a modal decomposer and an adaptive modal beamformer. The modal decomposer generates a plurality of zeroth-order eigenbeams from audio signals from an (e.g., spherical) array of audio sensors. The adaptive modal beamformer (i) steers the zeroth-order eigenbeams to a specified direction, (ii) adaptively generates a plurality of weighting coefficients for the plurality of zeroth-order eigenbeams, where the plurality of weighting coefficients satisfy a constraint of having only non-negative values, (iii) respectively applies the plurality of adaptively generated weighting coefficients to the plurality of steered, zeroth-order eigenbeams to generate a plurality of weighted, steered, zeroth-order eigenbeams, and (iv) combines the plurality of weighted, steered, zeroth-order eigenbeams to generate an output audio signal. Some embodiments have a further constraint that the weighting coefficients sum to a specified value (e.g., one).

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: 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-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.