A microphone array speech enhancement algorithm based on analysis/synthesis filtering that allows for variable signal distortion. The algorithm is used to suppress additive noise and interference. The processing structure consists of delaying the received signals so that the desired signal components add coherently, filtering each of the delayed signals through an analysis filter bank, summing the corresponding channel outputs from the sensors, applying a gain function to the channel outputs, and combining the weighted channel outputs using a synthesis filter. The structure uses two different gain functions, both of which are based on cross correlations of the channel signals from the two sensors. The first gain yields the GEQ-I array, which performs best for the case of a desired speech signal corrupted by uncorrelated white background noise. The second gain yields the GEQ-II array, which performs best for the case where there are more signals than microphones. The GEQ-II gain allows for a trade-off on a channel-dependent basis of additional signal degradation in exchange for additional noise and interference suppression.

Analysis/synthesis-based microphone array speech enhancer with variable signal distortion

Methods and systems for beamforming are disclosed that include a signal processor that can dynamically determine the relative time delays between a plurality of frequency-dependent signals. The signal processor can adaptively generate a beam signal by aligning the plural frequency-dependent signals according to the relative time delays between the signals. The signal processor can store one frequency-dependent signal as a reference signal and can align the remaining frequency-dependent signals relative to this reference signal. One advantage of the signal processor is that it can align the plural frequency-dependent signals generated by an array of microphones that can be arranged in a linear, two-dimensional or three-dimensional array and located in a room environment.

Methods and apparatus for adaptive beamforming

A telephone system includes two or more cardioid microphones held together and directed outwardly from a central point. Mixing circuitry and control circuitry combines and analyzes signals from the microphones and selects the signal from one of the microphones or from one of one or more predetermined combinations of microphone signals in order to track a speaker as the speaker moves about a room or as various speakers situated about the room speak then fall silent. Visual indicators, in the form of light emitting diodes (LEDs) are evenly spaced around the perimeter of a circle concentric with the microphone array. Mixing circuitry produces ten combination signals, A+B, A+C, B+C, A+B+C, A-B, B-C, A-C, A-0.5(B+C), B-0.5(A+C), and C-0.5(B+A), with the "listening beam" formed by combinations, such as A-0.5(B+C), that involve the subtraction of signals, generally being more narrowly directed than beams formed by combinations, such as A+B, that involve only the addition of signals. An omnidirectional combination A+B+C is employed when active speakers are widely scattered throughout the room. Weighting factors are employed in a known manner to provide unity gain output. Control circuitry selects the signal from the microphone or from one of the predetermined microphone combinations, based generally on the energy level of the signal, and employs the selected signal as the output signal. The control circuitry also operates to limit dithering between microphones and, by analyzing the beam selection pattern, may switch to a broader coverage pattern, rather than switching between two narrower beams that each covers one of the speakers.

Teleconferencing system which selects and optimal acoustic reception pattern from multiple patterns having different shapes and sizes

Stimuli in two or more sensory domains, such as an auditory domain and a visual domain, are combined in order to improve reliability and accuracy of detected user input. Detected events that occur substantially simultaneously in the multiple domains are deemed to represent the same user action, and if interpretable as a coherent action and provided to the system as interpreted input. The invention is applicable, for example, in a virtual keyboard or virtual controller, where stimuli resulting from user actions are detected, interpreted, and provided as input to a system.

Detecting, classifying, and interpreting input events based on stimuli in multiple sensory domains

A signal processing arrangement is connected to a microphone array to form at least one directable beam sound receiver. The directable beam sound receivers are adapted to receive sounds from predetermined locations in a prescribed environment such as auditorium. Signals representative of prescribed sound features received from the plurality of predetermined locations are generated and one or more of the locations is selected responsive to the sound feature signals. A plurality of directable beam sound receivers may be used to concurrently analyze sound features from the predetermined locations. Alternatively, one directable beam sound receiver may be used to scan the predetermined locations so that the sound feature signals therefrom are compared to sound features from a currently selected location.

Sound location arrangement

A circuit for reducing reverberative interference utilizes a pair of spatially separated microphones to obtain speech signals from a common sound source. Each speech signal is transformed into an envelope representative signal having rapid increases responsive to direct path and echo energy bursts from the sound source and exponential decaying portions between energy bursts. A first pulse corresponding to a sound source direct path energy burst is generated responsive to the first speech signal exceeding its envelope representative signal, and further first pulses corresponding to echo bursts are inhibited for a predetermined time. A second pulse corresponding to said sound source direct path energy burst is generated responsive to the second speech signal exceeding its envelope representative signal, and further second pulses corresponding to echo bursts are inhibited for a predetermined time. The first and second speech signals are aligned in phase responsive to the time difference between said first and second pulses. Three embodiments are disclosed: phase alignment by electronic delay adjustment using a pair of microphones or using vertical arrays of microphones, and phase alignment by feedback servo control of a rotatable microphone array.

Susan W. Christensen

Papers

Multiple microphone dereverberation system