Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: A method for managing an interaction of a calling party to a communication partner is provided. The method includes automatically determining if the communication partner expects DTMF input. The method also includes translating speech input to one or more DTMF tones and communicating the one or more DTMF tones to the communication partner, if the communication partner expects DTMF input.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: A novel beamforming post-processor technique with enhanced noise suppression capability. The present beam forming post-processor technique is a non-linear post-processing technique for sensor arrays (e.g., microphone arrays) which improves the directivity and signal separation capabilities. The technique works in so-called instantaneous direction of arrival space, estimates the probability for sound coming from a given incident angle or look-up direction and applies a time-varying, gain based, spatio-temporal filter for suppressing sounds coming from directions other than the sound source direction resulting in minimal artifacts and musical noise.

Abstract: A device for suppressing ambient sounds from speech received by a microphone array is provided. One embodiment of the device comprises a microphone array, a processor, an analog-to-digital converter, and memory comprising instructions stored therein that are executable by the processor.

Abstract: An omni-directional camera (a 360 degree camera) is proposed with an integrated microphone array. The primary application for such a camera is videoconferencing and meeting recording, and the device is designed to be placed on a meeting room table. The microphone array is in a planar configuration, and the microphones are located as close to the desktop as possible to eliminate sound reflections from the table. The camera is connected to the microphone array base with a thin cylindrical rod, which is acoustically invisible to the microphone array for the frequency range [50-4000] Hz. This provides a direct path from the person talking to all of the microphones in the array, and can therefore be used for sound source localization (determining the location of the talker) and beam-forming (improving the sound quality of the talker by filtering only sound from a particular direction). The camera array is elevated from the table to provide a near frontal viewpoint of the meeting participants.

Abstract: A system and process for dereverberation of multi-channel audio streams is presented which uses reverberation suppression techniques. In general, the present system and process builds a frequency dependent model of the reverberation decay and uses spectral subtraction-based reverberation reduction to achieve the aforementioned suppression. This dereverberation system and process can be used to improve automatic speech recognition (ASR) results with minimal CPU overhead.

Abstract: An “Interference Canceller” provides a computationally efficient real-time technique for removing stationary-tone interference from signals. Typical sources of stationary tone contamination of signals include noise from power wiring (i.e., 50/60 Hz or 400 Hz and their harmonics), frame or line frequencies from electronic devices, and noise from computer fans, hard disk drives, etc. In general, the Interference Canceller adaptively builds and updates a model of stationary tone interference in consecutive frames of an input signal. This adaptively updated model is then used to extrapolate and subtract noise from subsequent frames of the input signal to generate a “clean” output signal. This output signal exhibits significant attenuation of stationary tone interference without eliminating important portions of the underlying signal or distorting the underlying signal with artifacts such as musical noise or nonlinear distortions.

Abstract: Systems and methods are disclosed that facilitate real-time information exchange in a multimedia conferencing environment. Data Client(s) facilitate data collaboration between users and are maintained separately from audio/video (AV) Clients that provide real-time communication functionality. Data Clients can be remotely located with respect to one another and with respect to a server. A remote user Stand-in Device can be provided that comprises a display to present a remote user to local users, a digital automatic pan/tilt/zoom camera to capture imagery in, for example, a conference room and provide real-time information to an AV Client in a remote office, and a microphone array that can similarly provide real-time audio information from the conference room to an AV Client in the remote office. The invention further facilitates file transfer and presentation broadcast between Data Clients in a single location or in a plurality of disparate locations.

Abstract: A “Sensor Array Post-Filter” provides an adaptive post-filter that accurately models and suppresses both diffuse and directional noise sources as well as interfering speech sources. The post-filter is applied to an output signal produced by a beamformer used to process signals produced by a sensor array. As a result, the Sensor Array Post-Filter operates to improve the signal-to-noise ratio (SNR) of beamformer output signals by providing adaptive post-filtering of the output signals. The post-filter is generated based on a generative statistical model for modeling signal and noise sources at distinct regions in a signal field that considers prior distributions trained to model an instantaneous direction of arrival for signals captured by sensors in the array.

Abstract: A self-descriptive microphone array includes a microphone array memory, such as, for example a ROM, EEPROM, or other conventional memory, which contains a microphone array device description. This device description includes parametric information which defines operational characteristics and configuration of the microphone array. In further embodiments, the microphone array uses any of a variety of conventional wired or wireless computer interfaces, including serial, IEEE 1394, USB, Bluetooth™, etc., to connect to a computing device. Once connected, the microphone array provides its device description to the computing device. Sound processing software residing within the computing device is then automatically configured for optimally interacting with one or more analog or digital audio signals provided by the microphone array. In another embodiment, the microphone array performs integrated self calibration for automatically updating the device description.

Abstract: The presentation of location information to a user that is distracted by traveling can result in the user quickly forgetting, or never even comprehending, key parts of the location information, such as the street number. Identification can be made of intersections and points of interest near the user's destination, which can then be provided instead of, or in addition to, the address, thereby increasing user comprehension and retention, especially when distracted. Map data can be parsed into addresses, intersections and points of interest databases. These databases can be accessed to identify proximate intersections and points of interest, which can then be filtered and subsequently ranked to identify one intersection, one point of interest, or both, that can be presented to the user to aid the user in comprehending and retaining the location information even when distracted.

Abstract: Precision and reliability of localization estimates derived from conventional localization systems are improved through a system and method for post-processing of initial localization data, even in environments which may include noise, reflections, or other interference. Such localization systems include conventional sound source localization (SSL) systems based on microphone array inputs, radio source location systems based on directional antenna array inputs, etc. In general, this post-processing system and method applies statistical real-time clustering to initial localization estimates, and then uses this real-time clustering in a multi-stage process to generate new localization estimates having improved precision and reliability relative to the initial localization estimates.

Abstract: A novel beamforming post-processor technique with enhanced noise suppression capability. The present beam forming post-processor technique is a non-linear post-processing technique for sensor arrays (e.g., microphone arrays) which improves the directivity and signal separation capabilities. The technique works in so-called instantaneous direction of arrival space, estimates the probability for sound coming from a given incident angle or look-up direction and applies a time-varying, gain based, spatio-temporal filter for suppressing sounds coming from directions other than the sound source direction resulting in minimal artifacts and musical noise.

Abstract: A “Sensor Array Post-Filter” provides an adaptive post-filter that accurately models and suppresses both diffuse and directional noise sources as well as interfering speech sources. The post-filter is applied to an output signal produced by a beamformer used to process signals produced by a sensor array. As a result, the Sensor Array Post-Filter operates to improve the signal-to-noise ratio (SNR) of beamformer output signals by providing adaptive post-filtering of the output signals. The post-filter is generated based on a generative statistical model for modeling signal and noise sources at distinct regions in a signal field that considers prior distributions trained to model an instantaneous direction of arrival for signals captured by sensors in the array.

Abstract: An “Interference Canceller” provides a computationally efficient real-time technique for removing stationary-tone interference from signals. Typical sources of stationary tone contamination of signals include noise from power wiring (i.e., 50/60 Hz or 400 Hz and their harmonics), frame or line frequencies from electronic devices, and noise from computer fans, hard disk drives, etc. In general, the Interference Canceller adaptively builds and updates a model of stationary tone interference in consecutive frames of an input signal. This adaptively updated model is then used to extrapolate and subtract noise from subsequent frames of the input signal to generate a “clean” output signal. This output signal exhibits significant attenuation of stationary tone interference without eliminating important portions of the underlying signal or distorting the underlying signal with artifacts such as musical noise or nonlinear distortions.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: An analog preamplifier measurement system for a microphone array builds on conventional microphone arrays by providing an integral “self-calibration system.” This self-calibration system automatically injects an excitation pulse of a known magnitude and phase to all preamplifier inputs within the microphone array. The resulting analog waveform from each preamplifier output is then measured. A frequency analysis, such as, for example, a Fourier or Fast Fourier Transform (FFT), or other conventional frequency analysis, of each of the resulting waveforms is then performed. The results of this frequency analysis are then used to automatically compute frequency-domain compensation gains (e.g., magnitude and phase gains) for each preamplifier for matching or balancing the responses of all of the preamplifiers with each other.

Abstract: The ability to combine multiple audio signals captured from the microphones in a microphone array is frequently used in beamforming systems. Typically, beamforming involves processing the output audio signals of the microphone array in such a way as to make the microphone array act as a highly directional microphone. In other words, beamforming provides a “listening beam” which points to a particular sound source while often filtering out other sounds. A “generic beamformer,” as described herein automatically designs a set of beams (i.e., beamforming) that cover a desired angular space range within a prescribed search area. Beam design is a function of microphone geometry and operational characteristics, and also of noise models of the environment around the microphone array. One advantage of the generic beamformer is that it is applicable to any microphone array geometry and microphone type.

Abstract: A system and process for sound source localization (SSL) utilizing beamsteering is presented. The present invention provides for improved beamsteering with less drain on system resources while providing accurate, real time results. To accomplish this, the present SSL system and process rejects as much as possible extraneous audio frames and analyzes only those frames exhibiting a well defined sound source. In addition, the number of beams is reduced as much as possible to save on processing time, but a full scan of the working volume is still made with the beams. And finally, interpolation is used to increase the precision of the technique.