Abstract: Area tracking systems including a plurality of short range sensors that are communicatively connected to a server and methods of tracking electronic devices. The electronic devices are tracked by polling for the electronic devices, the polling including transmission of electronic addresses corresponding to the electronic devices and determining whether a reply to the poling is received. The electronic addresses are each stored on one of a plurality of lists. Each of the lists may be polled at a different frequency and may include a different number of electronic device addresses.

Abstract: A method and apparatus for performing active speaker selection in teleconferencing applications illustratively comprises a microphone array module, a speaker recognition system, a user interface, and a speech signal selection module. The microphone array module separates the speech signal from each active speaker from those of other active speakers, providing a plurality of individual speaker's speech signals. The speaker recognition system identifies each currently active speaker using conventional speaker recognition/identification techniques. These identities are then transmitted to a remote teleconferencing location for display to remote participants via a user interface.

Abstract: Area tracking systems including a plurality of short range sensors that are communicatively connected to a server and methods of tracking electronic devices. The electronic devices are tracked by polling for the electronic devices, the polling including transmission of electronic addresses corresponding to the electronic devices and determining whether a reply to the poling is received. The electronic addresses are each stored on one of a plurality of lists. Each of the lists may be polled at a different frequency and may include a different number of electronic device addresses.

Abstract: A telestration system comprises a telestration server configured to communicate with an arbitrary number of telestration clients over a network. The telestration server is further configured to receive telestration input signals from respective ones of the clients and to send telestration output signals to the respective clients, with the telestration output signal sent to a given one of the clients comprising telestration information derived from the telestration input signal received from at least one other one of the clients. Each of the clients with the support of the telestration server can generate a combined telestration overlay for presentation with associated visual content shared between the clients.

Abstract: A method for performing packet loss concealment of lost packets in Voice over IP (Internet Protocol) speech transmission. Statistical n-gram models are initially created with use of a training speech corpus, and then, packets lost during transmission are advantageously replaced based on these models. In particular, the existence of statistical patterns in successive voice over IP (VoIP) packets is advantageously exploited by first using conventional vector quantization (VQ) techniques to quantize the parameter data for each packet with use of a corresponding VQ index, and then determining statistical correlations between consecutive sequences of such VQ indices representative of the corresponding sequences of n packets. The statistic n-gram predictive models so created are then used to predict parameter data for use in representing lost data packets.

Abstract: A method and apparatus for performing active speaker selection in teleconferencing applications illustratively comprises a microphone array module, a speaker recognition system, a user interface, and a speech signal selection module. The microphone array module separates the speech signal from each active speaker from those of other active speakers, providing a plurality of individual speaker's speech signals. The speaker recognition system identifies each currently active speaker using conventional speaker recognition/identification techniques. These identities are then transmitted to a remote teleconferencing location for display to remote participants via a user interface.

Abstract: A method and apparatus for interactive language instruction is provided that displays text files for processing, provide key features and functions for interactive learning, displays facial animation, and provides a workspace for language building functions. The system includes a stored set of language rules as part of the text-to-speech sub-system, as well as another stored set of rules as applied to the process of learning a language. The method implemented by the system includes digitally converting text to audible speech, providing the audible speech to a user or student (with the aid of an animated image in selected circumstances), prompting the student to replicate the audible speech, comparing the student's replication with the audible speech provided by the system, and providing feedback and reinforcement to the student by, for example, selectively recording or playing back the audible speech and the student's replication.

Abstract: A method for performing packet loss concealment of lost packets in Voice over IP (Internet Protocol) speech transmission. Statistical n-gram models are initially created with use of a training speech corpus, and then, packets lost during transmission are advantageously replaced based on these models. In particular, the existence of statistical patterns in successive voice over IP (VoIP) packets is advantageously exploited by first using conventional vector quantization (VQ) techniques to quantize the parameter data for each packet with use of a corresponding VQ index, and then determining statistical correlations between consecutive sequences of such VQ indices representative of the corresponding sequences of n packets. The statistic n-gram predictive models so created are then used to predict parameter data for use in representing lost data packets.

Abstract: A method and apparatus for performing real-time endpoint detection for use in automatic speech recognition. A filter is applied to the input speech signal and the filter output is then evaluated with use of a state transition diagram (i.e., a finite state machine). The filter is advantageously designed in light of several criteria in order to increase the accuracy and robustness of detection. The state transition diagram advantageously has three states. The endpoints which are detected may then be advantageously applied to the problem of energy normalization of the speech portion of the signal.

Abstract: A method and apparatus for interactive language instruction is provided that displays text files for processing, provide key features and functions for interactive learning, displays facial animation, and provides a workspace for language building functions. The system includes a stored set of language rules as part of the text-to-speech sub-system, as well as another stored set of rules as applied to the process of learning a language. The method implemented by the system includes digitally converting text to audible speech, providing the audible speech to a user or student (with the aid of an animated image in selected circumstances), prompting the student to replicate the audible speech, comparing the student's replication with the audible speech provided by the system, and providing feedback and reinforcement to the student by, for example, selectively recording or playing back the audible speech and the student's replication.

Abstract: A method and apparatus for performing real-time endpoint detection for use in automatic speech recognition. A filter is applied to the input speech signal and the filter output is then evaluated with use of a state transition diagram (i.e., a finite state machine). The filter is advantageously designed in light of several criteria in order to increase the accuracy and robustness of detection. The state transition diagram advantageously has three states. The endpoints which are detected may then be advantageously applied to the problem of energy normalization of the speech portion of the signal.

Abstract: Automatic Speech Recognition (ASR) is achieved in wireless communications systems in which reliable ASR feature vector sequences are derived at a base station directly from digitally transmitted speech coder parameters, with no additional processing or signal modification required at the originating handset. No secondary channel need be provided for the transmission of ASR feature vectors. In operating on received speech coder parameters prior to conversion to a voice signal the present system and methods avoid the lossy conversion process and associated voice distortion. Since the received voice parameters are error protected during transmission they are received with greater accuracy. All, or a subset, of speech coding parameters, including, in appropriate cases, spectral envelope parameters, reflection coefficients, LSPs, LSFs, LPCs, LPCCs, and weighted LPCCs may be processed at a receiving base station or forwarded to another location for processing.