Abstract: A method and an apparatus for using a plurality of antennas to equalize a composite propagation channel of a wideband wireless communication is provided. The method comprises adjusting a corresponding weight associated with at least two of the plurality of antennas to obtain a channel response from the plurality of antennas over a composite propagation channel. The method further comprises equalizing the channel response of the composite propagation channel over a given bandwidth based on the corresponding weights for the wireless communication over a wideband frequency range. By adjusting a complex weight associated with each of the set of transmit antennas, a wireless communication system may obtain a channel response from the plurality of antennas over the composite propagation channel.

Abstract: An endpoint or other communication device of a communication system includes a clock recovery module. The communication device is operative as a slave device relative to another communication device that is operative as a master device. The clock recovery module comprises a clock recovery loop configured to control a slave clock frequency of the slave device so as to synchronize the slave clock frequency with a master clock frequency of the master device. The clock recovery loop utilizes a frequency error estimator implemented as a maximum-likelihood estimator with slope fitting based on a sequence of arrival timestamps, and a loop filter implemented as a series combination of an adaptive-bandwidth filter and a proportional-integral controller.

Abstract: An endpoint or other communication device of a communication system includes a clock recovery module. The communication device is operative as a slave device relative to another communication device that is operative as a master device. The clock recovery module comprises a clock recovery loop configured to control a slave clock frequency of the slave device so as to synchronize the slave clock frequency with a master clock frequency of the master device. The clock recovery loop comprises a primary loop having a first frequency error estimator for generating a first estimate of error between the master and slave clock frequencies, a second frequency error estimator outside of the primary loop for generating a second estimate of error between the master and slave clock frequencies, and an accumulator coupled between the second frequency error estimator and the primary loop. The second estimate is controllably injected into the primary loop via the accumulator.

Abstract: An endpoint or other communication device of a communication system includes a clock recovery module. The communication device is operative as a slave device relative to another communication device that is operative as a master device. The clock recovery module comprises a clock recovery loop configured to control a slave clock frequency of the slave device so as to synchronize the slave clock frequency with a master clock frequency of the master device. The clock recovery loop utilizes a frequency error estimator implemented as a maximum-likelihood estimator with slope fitting based on a sequence of arrival timestamps, and a loop filter implemented as a series combination of an adaptive-bandwidth filter and a proportional-integral controller.

Abstract: An endpoint or other communication device of a communication system includes a clock recovery module. The communication device is operative as a slave device relative to another communication device that is operative as a master device. The clock recovery module comprises a clock recovery loop configured to control a slave clock frequency of the slave device so as to synchronize the slave clock frequency with a master clock frequency of the master device. The clock recovery loop comprises a primary loop having a first frequency error estimator for generating a first estimate of error between the master and slave clock frequencies, a second frequency error estimator outside of the primary loop for generating a second estimate of error between the master and slave clock frequencies, and an accumulator coupled between the second frequency error estimator and the primary loop. The second estimate is controllably injected into the primary loop via the accumulator.

Abstract: The invention is a method and apparatus for frequency-domain adaptive filtering that has broad applications such as to equalizers, but is particularly suitable for use in acoustic echo cancellation circuits for stereophonic and other multiple channel teleconferencing systems. The method and apparatus utilizes a frequency-domain recursive least squares criterion that minimizes the error signal in the frequency-domain. In order to reduce the complexity of the algorithm, a constraint is removed resulting in an unconstrained frequency-domain recursive least mean squares method and apparatus. A method and apparatus for selecting an optimal adaptation step for the UFLMS is disclosed. The method and apparatus is generalized to the multiple channel case and exploits the cross-power spectra among all of the channels.

Abstract: In a transmitter, an upconverter converts a lower frequency signal to a higher frequency signal. Prior to the upconversion, a compensator compensates for at least gain/phase distortion that will be introduced into the lower frequency signal by at least the upconverter.

Abstract: The invention is a method and apparatus for frequency-domain adaptive filtering that has broad applications such as to equalizers, but is particularly suitable for use in acoustic echo cancellation circuits for stereophonic and other multiple channel teleconferencing systems. The method and apparatus utilizes a frequency-domain recursive least squares criterion that minimizes the error signal in the frequency-domain. In order to reduce the complexity of the algorithm, a constraint is removed resulting in an unconstrained frequency-domain recursive least mean squares method and apparatus. A method and apparatus for selecting an optimal adaptation step for the UFLMS is disclosed. The method and apparatus is generalized to the multiple channel case and exploits the cross-power spectra among all of the channels.

Abstract: The invention is a method and apparatus for frequency-domain adaptive filtering that has broad applications such as to equalizers, but is particularly suitable for use in acoustic echo cancellation circuits for stereophonic and other multiple channel teleconferencing systems. The method and apparatus utilizes a frequency-domain recursive least squares criterion that minimizes the error signal in the frequency-domain. In order to reduce the complexity of the algorithm, a constraint is removed resulting in an unconstrained frequency-domain recursive least mean squares method and apparatus. A method and apparatus for selecting an optimal adaptation step for the UFLMS is disclosed. The method and apparatus is generalized to the multiple channel case and exploits the cross-power spectra among all of the channels.

Abstract: A signal is predistorted by producing a set of sample values, each of at least a subset of which is dependent on (i) at least one of a plurality of past time spaced input samples and (ii) a current time spaced input sample, and independent of any other time spaced input sample, and combining the sample values to produce the predistorted signal. Predistortion circuitry for generating the predistorted signal may be implemented using multiple predistortion core circuits, with each of the predistortion core circuits receiving a data input and an index input associated with a particular input sample and generating a corresponding data output. The data outputs of the predistortion core circuits correspond generally to sample values. The predistortion circuitry may also include at least one memory finite impulse response (FIR) filter which processes one or more input samples in conjunction with the production of the sample values.

Abstract: A signal is predistorted by producing a first set of sample values using a first predistortion technique, producing a second set of sample values using a second predistortion technique different than the first predistortion technique, and combining the first and second sets of sample values to produce a predistorted signal. At least one of the first and second predistortion techniques produce the corresponding set of sample values based at least in part on a plurality of past time spaced input samples relative to a current time spaced input sample.

Abstract: A method for processing signals from multiple receiving/transmitting elements that define an array antenna is disclosed. The method includes measuring uplink signals from the elements of the array antenna to form a directional downlink antenna response. A first covariance matrix between pairs of uplink signals is determined for desired signal components of the measured uplink signals. A second covariance matrix is determined for noise and interference components of the measured uplink signals. An antenna processing system computes adaptive weights based on the first covariance matrix, the second covariance matrix and a loading term to in effect form a directional downlink radiation pattern. The loading term represents a balance adjustment between a first performance rating of a particular coverage area versus second performance rating of the wireless communication system.

Abstract: A signal is predistorted by producing a first set of sample values using a first predistortion technique, producing a second set of sample values using a second predistortion technique different than the first predistortion technique, and combining the first and second sets of sample values to produce a predistorted signal. At least one of the first and second predistortion techniques produce the corresponding set of sample values based at least in part on a plurality of past time spaced input samples relative to a current time spaced input sample.

Abstract: A signal is predistorted by producing a set of sample values, each of at least a subset of which is dependent on (i) at least one of a plurality of past time spaced input samples and (ii) a current time spaced input sample, and independent of any other time spaced input sample, and combining the sample values to produce the predistorted signal. Predistortion circuitry for generating the predistorted signal may be implemented using multiple predistortion core circuits, with each of the predistortion core circuits receiving a data input and an index input associated with a particular input sample and generating a corresponding data output. The data outputs of the predistortion core circuits correspond generally to sample values. The predistortion circuitry may also include at least one memory finite impulse response (FIR) filter which processes one or more input samples in conjunction with the production of the sample values.

Abstract: A method and apparatus for estimating individual impulse responses for a stereophonic communication system, such as a teleconferencing system, which involves selectively reducing the correlation between the individual channel signals of the stereophonic system. Selective reduction of stereophonic source signal correlation advantageously results in the estimation of individual impulse responses of a receiving room of the stereophonic communication system. The selectively reduced-correlation source signals are provided to conventional adaptive filters and the receiving room loudspeakers. Automatic echo cancellation is performed in a conventional fashion, but on the selectively reduced-correlation source signals. Specifically, selective reduction of source signal correlation between two stereophonic channels of a teleconferencing system is achieved by introducing a small non-linearity into each channel in order to reduce the interchannel coherence.