| Patent application number | Description | Published |
|---|
| 20080247565 | Position-Independent Microphone System - 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. | 10-09-2008 |
| 20080260175 | Dual-Microphone Spatial Noise Suppression - 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. | 10-23-2008 |
| 20090175466 | NOISE-REDUCING DIRECTIONAL MICROPHONE ARRAY - 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. | 07-09-2009 |
| 20090316916 | Wide Dynamic Range Microphone - 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. Methods of operating the system may include equalizing the outputs of two or more transducers prior to coupling one or more outputs to the system output. | 12-24-2009 |
| 20100008517 | AUDIO SYSTEM BASED ON AT LEAST SECOND-ORDER EIGENBEAMS - 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. | 01-14-2010 |
| 20100202628 | AUGMENTED ELLIPTICAL MICROPHONE ARRAY - 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. | 08-12-2010 |
