Abstract: A route can be optimized according to various constraints, specifically towards non-travel constraints. A number of community activities can take place such that the activities impact traffic. Thus, a route can be produces that lowers an influence the activities have upon a route. Moreover, a route can be produced that improves signal strength, such as strength of a signal used to communicate with a cellular telephone. In addition, a route can be enhanced by taking into account weather conditions, including real-time weather in addition to weather predictions.

Abstract: A user can be compensated for taking detours from a projected route. Commonly, the reason for the compensation is that the user will be subjected to advertising, the user will pass by an establishment she is likely to visit, or to ease traffic congestion. Analysis of an area takes place and monetization opportunities are determined based upon the results of the analysis. A route between at least about two locations can be altered such that the user is provided a reward, commonly in an optimized manner.

Abstract: A microphone array having at least three microphones provides a captured signal. Spatial noise suppression estimates a desired signal from a captured signal using spatio-temporal distribution of the speech and the noise. In particular, spatial information indicative of at least two quantities of direction are used. A first quantity is based on a first combination of the signals from the at least three microphones, a second quantity is based on a second combination of the signals of the at least three microphones.

Abstract: A system and process for self calibrating a plurality of audio sensors of a microphone array on a continuous basis, while the array is in operation, is presented. In essence, the present microphone array self calibration system and process finds a set of corrective gains that provides the best channel matching amongst the audio sensors of the array by compensating for the differences in the sensor parameters. The present system and process is not CPU use intensive and is capable of providing real-time microphone array self-calibration. It is based on a simplified channel model, projection of sensor coordinates on the direction of arrival (DOA) line, and approximation of received energy levels, all of which speed up processing time.

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: 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: 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: 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: 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: 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 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: 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: A system and process for self calibrating a plurality of audio sensors of a microphone array on a continuous basis, while the array is in operation, is presented. In essence, the present microphone array self calibration system and process finds a set of corrective gains that provides the best channel matching amongst the audio sensors of the array by compensating for the differences in the sensor parameters. The present system and process is not CPU use intensive and is capable of providing real-time microphone array self-calibration. It is based on a simplified channel model, projection of sensor coordinates on the direction of arrival (DOA) line, and approximation of received energy levels, all of which speed up processing time.

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.

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.