Abstract: Certain exemplary embodiments provide a method, comprising: automatically detecting at an agent an installation of a communications device to a local network coupled to the agent; automatically offering via the communications device a plurality of user-selectable service options relating to a service provider's network and relating to at least one capability of the communications device, the at least one capability detected by the agent; in response to a selection of at least one service option from the plurality of service options, providing via the agent a user connection address.

Abstract: Certain exemplary embodiments provide a method, comprising: automatically detecting at an agent an installation of a communications device to a local network coupled to the agent; automatically offering via the communications device a plurality of user-selectable service options relating to a service provider's network and relating to at least one capability of the communications device, the at least one capability detected by the agent; in response to a selection of at least one service option from the plurality of service options, providing via the agent a user connection address.

Abstract: Certain exemplary embodiments provide a method, comprising: automatically detecting at an agent an installation of a communications device to a local network coupled to the agent; automatically offering via the communications device a plurality of user-selectable service options relating to a service provider's network and relating to at least one capability of the communications device, the at least one capability detected by the agent; in response to a selection of at least one service option from the plurality of service options, providing via the agent a user connection address.

Abstract: An algorithm for computing an efficient, reduced complexity, windowed optimal linear time domain equalizer for a dispersive channel comprises the steps of determining a window of maximum energy in the impulse response of length equal to or less than a number of cyclic prefix samples associated with a received digital data signal, computing the corresponding inside and outside matrices, performing an inverse Cholesky decomposition of the inside matrix, creating a resultant matrix as the product of the outer and the upper and lower square root inner matrix, followed by Householder reduction and QL transformation to thereby compute the time domain equalizer as the linear transformation of the eigenvector corresponding to the smallest eigenvalue at the receiver. The smallest eigenvalue is determined using the aforementioned orthogonal transformations without determining all the eigenvalues efficiently but without the loss accuracy associated with iterative methods like the conventional power method.

Abstract: An algorithm for computing an efficient, reduced complexity, windowed optimal linear time domain equalizer for a dispersive channel comprises the steps of determining a window of maximum energy in the impulse response of length equal to or less than a number of cyclic prefix samples associated with a received digital data signal, computing the corresponding inside and outside matrices, performing an inverse Cholesky decomposition of the inside matrix, creating a resultant matrix as the product of the outer and the upper and lower square root inner matrix, followed by Householder reduction and QL transformation to thereby compute the time domain equalizer as the linear transformation of the eigenvector corresponding to the smallest eigenvalue at the receiver. The smallest eigenvalue is determined using the aforementioned orthogonal transformations without determining all the eigenvalues efficiently but without the loss accuracy associated with iterative methods like the conventional power method.

Abstract: An algorithm for computing an efficient, reduced complexity, windowed optimal linear time domain equalizer for a dispersive channel comprises the steps of determining a window of maximum energy in the impulse response of length equal to or less than a number of cyclic prefix samples associated with a received digital data signal, computing the corresponding inside and outside matrices, performing an inverse Cholesky decomposition of the inside matrix, creating a resultant matrix as the product of the outer and the upper and lower square root inner matrix, followed by Householder reduction and QL transformation to thereby compute the time domain equalizer as the linear transformation of the eigenvector corresponding to the smallest eigenvalue at the receiver. The smallest eigenvalue is determined using the aforementioned orthogonal transformations without determining all the eigenvalues efficiently but without the loss accuracy associated with iterative methods like the conventional power method.

Abstract: An algorithm for computing an efficient, reduced complexity, windowed optimal linear time domain equalizer for a dispersive channel comprises the steps of determining a window of maximum energy in the impulse response of length equal to or less than a number of cyclic prefix samples associated with a received digital data signal, computing the corresponding inside and outside matrices, performing an inverse Cholesky decomposition of the inside matrix, creating a resultant matrix as the product of the outer and the upper and lower square root inner matrix, followed by Householder reduction and QL transformation to thereby compute the time domain equalizer as the linear transformation of the eigenvector corresponding to the smallest eigenvalue at the receiver. The smallest eigenvalue is determined using the aforementioned orthogonal transformations without determining all the eigenvalues efficiently but without the loss accuracy associated with iterative methods like the conventional power method.

Abstract: A method is presented for increasing data throughput in a communications channel comprising the steps of selecting a first data packet from a queue of data packets to be transmitted over a communications channel, modulating said first data packet using a first modulation scheme, selecting a second data packet from a queue of data packets to be transmitted over the communications channel, modulating the second data packet using a second modulation scheme overlaying the first data packet on a symbol by symbol basis, transmitting the first data packet overlaid with second data packet over the communications channel, determining whether the first data packet was received by monitoring using the first modulation scheme for an acknowledgment for the first data packet before expiration of a timeout period, determining whether the second data packet was received by monitoring using the first modulation scheme for an acknowledgment for the second data packet before expiration of the timeout period and repeating the st

Abstract: A method is presented for increasing data throughput in a communications channel comprising the steps of selecting a first data packet from a queue of data packets to be transmitted over a communications channel, modulating said first data packet using a first modulation scheme, selecting a second data packet from a queue of data packets to be transmitted over the communications channel, modulating the second data packet using a second modulation scheme overlaying the first data packet on a symbol by symbol basis, transmitting the first data packet overlaid with second data packet over the communications channel, determining whether the first data packet was received by monitoring using the first modulation scheme for an acknowledgment for the first data packet before expiration of a timeout period, determining whether the second data packet was received by monitoring using the first modulation scheme for an acknowledgment for the second data packet before expiration of the timeout period and repeating the st

Abstract: An algorithm for computing an efficient, reduced complexity, windowed optimal linear time domain equalizer for a dispersive channel comprises the steps of determining a window of maximum energy in the impulse response of length equal to or less than a number of cyclic prefix samples associated with a received digital data signal, computing the corresponding inside and outside matrices, performing an inverse Cholesky decomposition of the inside matrix, creating a resultant matrix as the product of the outer and the upper and lower square root inner matrix, followed by Householder reduction and QL transformation to thereby compute the time domain equalizer as the linear transformation of the eigenvector corresponding to the smallest eigenvalue at the receiver. The smallest eigenvalue is determined using the aforementioned orthogonal transformations without determining all the eigenvalues efficiently but without the loss accuracy associated with iterative methods like the conventional power method.

Abstract: A new communication system merges a voice communication with an image communication into a single transmission channel to provide a virtual voice/video communication service over a limited channel bandwidth. A voice communication path is established between two subscriber communication stations and a desired still picture image is transmitted via the voice channel at the start of the transmission. Portions of the still image expected to be visually active during the communication, such as facial movements (e.g. the lips of the speaker), are replaced at the receiver by an idealized representative image. The voice portion of the transmission is digitally encoded and transmitted from the transmitting station to the receiving station. Synchronous activation of the idealized image (e.g. the lips) are animated by code book coefficients of the digitized speech message.

Abstract: A personal communicator handset is designed to operate as both a handset and as a speakerphone and to automatically switch between the two modes based on distance between the handset and the user's ear. This distance is determined by an infrared range detection unit built into the handset.

Abstract: An acoustic calibration circuit in a voice switched adaptive speakerphone accurately determines the type of acoustic environment of both large and small rooms having either harsh or favorable acoustics. The calibration circuit determines the acoustic environment of a room by emitting a tone burst through a loudspeaker associated with the speakerphone and generating an echo decay parameter that is indicative of the duration of echoes from the tone burst signal measured with a microphone also associated with the speakerphone. The echo decay parameter is generated from a composite representation of the acoustic response for those echoes having the largest returned amplitude level measued at each one of multiple predetermined time intervals by the calibration circuit.

Abstract: An improved switched-loss, adaptive speakerphone dynamically adjusts its switching thresholds and other performance parameters based on an analysis of acoustic environment and telephone line conditions. To access these conditions, the speakerphone utilizes a computer with associated firmware. As part of a calibration process, the speakerphone computes its thresholds before each use to compensate for possible variations in hardware circuitry therein. This is achieved by passing a first and then a second level of a test tone signal through the hardware circuitry and measuring the resulting response. The speakerphone also measures the acoustics of the room in which it operates as part of the calibration process. For this measurement, a tone burst signal is generated through a loudspeaker in the speakerphone and measured by a microphone also in the speakerphone. A time-domain acoustic response of the room is obtained which provides to the computer the amplitude of the return signal and the duration of its echo.

Abstract: An adaptive speakerphone employs an echo suppression arrangement which compensates for a reverberant effect that occurs during operation of the speakerphone in a near full or full duplex mode. This reverberant effect allows the far-end party to hear his or her own echo from the near-end location. The arrangement includes an echo suppression process which compensates for this effect by inserting loss in the transmit path of the speakerphone as appropriate. Operation of this process is such that whenever the speakerphone is in the receive state, the levels of a transmit and receive speech signal in the speakerphone are monitored. By monitoring these speech signal levels, the process determines when speech into a microphone in the speakerphone is being coupled from a loudspeaker also in the speakerphone and when the transmit speech signal level exceeds a predetermined threshold. If this threshold is exceeded, then loss that tracks an envelope of the receive speech signal is inserted in the transmit path.

Abstract: An acoustic calibration circuit in a voice switched adaptive speakerphone accurately determines the type of acoustic environment in which the speakerphone is employed. The calibration circuit measures the acoustics of the room by emitting a tone burst through a loudspeaker associated with the speakerphone and measuring the returned time-domain acoustic response with a microphone also associated with the speakerphone. Obtained from this response and processed by a computer in the speakerphone are the maximum amplitude of the returned signal, and the duration of the echoes. The amplitude of the returned signal determines what level of transmit speech will be required to break in on receive speech. The greater the acoustic return, the higher that threshold must be to protect against self-switching. And the duration of the echoes determine how quickly speech energy injected into the room will dissipate, which, in turn, controls how fast the speakerphone can switch from a receive to a transmit state.

Abstract: A computer controlled speakerphone includes a line adapting arrangement for developing information about an interface between the speakerphone and a hybrid in a communication line and for providing optimal performance during operation by adapting operating parameters of the speakerphone to the line. During a conversation, the line adapting arrangement measures and averages the degree of hybrid reflection that is presented to the speakerphone. This hybrid reflection provides a measure of both the hybrid and a far-end acoustic return. By determining the degree of hybrid reflection, the switching threshold level of the speakerphone for switching between the transmit state and the receive state may be adjusted. Once the expected level of receive speech due to hybrid reflection is known, additional receive speech due to the far-end talker may be accurately determined and the state of the speakerphone switched accordingly.

Abstract: An electrical calibration circuit in a voice switched adaptive speakerphone accurately determines the operating parameters of the speakerphone for optimal performance before operation. This is achieved by updating thresholds used to determine when the speakerphone should be in each of three possible operating states; transmit, receive or idle. During operation, these thresholds are compared with transmit and receive signal strength levels to determine the state of the speakerphone. The thresholds are updated to counteract the effects of variation and aging of hardware circuits in the speakerphone and are obtained by having the calibration circuit generate a test tone signal which is passed through speech processing circuitry in the speakerphone at two different levels and the resulting response measured.