Abstract: An embodiment of the present invention is a method and apparatus for extending bandwidth of a speech communication beyond a band-limited region to which the speech communication may be otherwise constrained. Such embodiments may be used to provide higher fidelity speech to the listener for an enhanced user experience.

Abstract: Methods and apparatus are disclosed to extend the bandwidth of a speech communication to yield a perceived higher quality speech communication for an enhanced user experience. In one aspect of the invention, for example, methods and apparatus can be used to extend the bandwidth of a speech communication beyond a band-limited region defined by the lowest limit and highest limit of the frequency spectrum by which such speech communication is otherwise characterized absent such bandwidth extension. In another aspect of the invention, for example, methods and apparatus can be used to substitute for corrupt, missing or lost components of a given speech communication, or to otherwise enhance the perceived quality of a speech communication, by extending the speech communication to include one or more artificially created points within the region defined by the lowest limit and highest limit of the frequency spectrum by which such speech communication is characterized.

Abstract: A technique for interference level estimation for a WCDMA system based on the low order statistics of Beaulieu Series is provided. Correlation of a locally generated copy of a dedicated physical control channel (DPCCH) with the input samples of the radio frequency message is performed using low order statistics of a Beaulieu Series to estimate an interference level and thereby implicitly separate a set of multipath components from a noise floor.

Abstract: Methods and apparatus to extend the bandwidth of a speech communication to yield a perceived higher quality speech communication for an enhanced user experience. In one aspect of the invention, for example, methods and apparatus can be used to extend the bandwidth of a speech communication beyond a band-limited region defined by the lowest limit and highest limit of the frequency spectrum by which such speech communication is otherwise characterized absent such bandwidth extension. In another aspect of the invention, for example, methods and apparatus can be used to substitute for corrupt, missing or lost components of a given speech communication, or to otherwise enhance the perceived quality of a speech communication, by extending the speech communication to include one or more artificially created points within the region defined by the lowest limit and highest limit of the frequency spectrum by which such speech communication is characterized.

Abstract: A method for preamble detection in mobile unit to base unit wireless telephony sets a preamble detection threshold based upon a Beaulieu series computation dependent upon preamble correlation data. This preamble detection threshold adjusts for noise by assuming the noise is additive White Gaussian noise (AWGN) with a known variance. The method determines the threshold for achieving a probability of false detection of the preamble from noise input of less than 0.001.

Abstract: A transform-domain adaptive filter uses selective partial updating of adaptive filter parameters. This updating may be based on a constrained minimization problem. The adaptive filter parameters are separated into subsets, and a subset is selected to be updated at each iteration. A normalization process applied to the frequency bins prior to multiplication by the adaptive filter parameters is used to prevent adaptive filter lock-up that may be experienced in the event of high energy levels of signals in particular frequency bins. Convergence of the transform domain filter is ensured at a rate generally faster than a corresponding time-domain adaptive filter. The transform-domain adaptive filter may be used for various applications, including system identification, channel equalization, or echo cancellation.

Abstract: Detectors determine the presence of valid sinusoids for DTMF, MF-R1 and MF-R2 protocols for encoding dialed digits. The detectors split electrical signals into subbands. Energies within the subbands are analyzed to determine a presence of sinusoids corresponding to frequencies of dialed digits. In one embodiment, the detectors comprise a PS-IIR filter to split the electrical signal into the subbands. The detectors further comprise at least one bank of filters, such as notch filters, corresponding to the number of possible relevant frequencies within the respective subbands. The detectors further comprise detection logic comprising tests, which may include analyzing the output(s) from the bank of filters. Optionally, a preclassifier is employed to predetermine which filters in the banks of filters are to be executed. The detectors, typically deployed in digital signal processors, allow for an increase in the density of detectors and provide robust performance in talk-off situations.

Abstract: A transform-domain adaptive filter uses selective partial updating of adaptive filter parameters. This updating may be based on a constrained minimization problem. The adaptive filter parameters are separated into subsets, and a subset is selected to be updated at each iteration. A normalization process applied to the frequency bins prior to multiplication by the adaptive filter parameters is used to prevent adaptive filter lock-up that may be experienced in the event of high energy levels of signals in particular frequency bins. Convergence of the transform domain filter is ensured at a rate generally faster than a corresponding time-domain adaptive filter. The transform-domain adaptive filter may be used for various applications, including system identification, channel equalization, or echo cancellation.

Abstract: A non-linear processor for managing a speech signal is disclosed. The non-linear processor includes an adaptive lattice filter that generates a prediction error from a residual echo of the speech signal, a signum non-linearity unit that generates an excitation component from the prediction error, an excitation signal mixer that generates an excitation signal from the excitation component and a noise component, an inverse lattice filter that generates a modified residual echo from the excitation signal, and a gain adjuster that adjusts a power of the modified residual echo.

Abstract: A non-linear processor for managing a speech signal is disclosed. The non-linear processor includes an adaptive lattice filter that generates a prediction error from a residual echo of the speech signal, a signum non-linearity unit that generates an excitation component from the prediction error, an excitation signal mixer that generates an excitation signal from the excitation component and a noise component, an inverse lattice filter that generates a modified residual echo from the excitation signal, and a gain adjuster that adjusts a power of the modified residual echo.

Abstract: A method for preamble detection in mobile unit to base unit wireless telephony sets a preamble detection threshold based upon a Beaulieu series computation dependent upon preamble correlation data. This preamble detection threshold adjusts for noise by assuming the noise is additive White Gaussian noise (AWGN) with a known variance. The method determines the threshold for achieving a probability of false detection of the preamble from noise input of less than 0.001.

Abstract: The present invention is a method and apparatus for estimating the bulk delay and span associated with an echo signal on a telephone line carrying a proximal signal and an echo signal. A baseline solution is to use the echo canceller itself to estimate the bulk-delay. After achieving a reasonable level of echo cancellation, the bulk-delay and the echo-span are estimated from the adaptive coefficients, followed by an enhanced mode of operation using the estimated parameters. Alternatively, for longer, unknown bulk delays, a subband adaptive system is used to estimate the bulk-delay and echo-span. The subband adaptive echo canceller outputs the set of subband adaptive coefficients to a full-band echo canceller after achieving a reasonable level of convergence.

Abstract: A reduced-complexity, fast converging adaptive filter may be used for network echo cancellation applications, including applications having sparse echo paths. The new filter, referred to as a selective-partial-update proportionate NLMS filter, may be based on a proportionate NLMS (PNLMS) technique and selective partial updating of the adaptive filter coefficients. The new PNLMS filter may exploit sparseness of a communications channel to speed up the initial convergence of the NLMS technique included in the filter by weighting regressor data proportionately with an estimated magnitude of the channel impulse response. Selective partial updating is essentially a data selection method to reduce the computational complexity. The performance of the selective-partial-update PNLMS filter compares favorably to an adaptive filter using standard PNLMS for echo paths specified in ITU-T Recommendation G.168.

Abstract: A transform-domain adaptive filter uses selective partial updating of adaptive filter parameters. This updating may be based on a constrained minimization problem. The adaptive filter parameters are separated into subsets, and a subset is selected to be updated at each iteration. A normalization process applied to the frequency bins prior to multiplication by the adaptive filter parameters is used to prevent adaptive filter lock-up that may be experienced in the event of high energy levels of signals in particular frequency bins. Convergence of the transform domain filter is ensured at a rate generally faster than a corresponding time-domain adaptive filter. The transform-domain adaptive filter may be used for various applications, including system identification, channel equalization, or echo cancellation.

Abstract: A non-linear processor for managing a speech signal is disclosed. The non-linear processor includes an adaptive lattice filter that generates a prediction error from a residual echo of the speech signal, a signum non-linearity unit that generates an excitation component from the prediction error, an excitation signal mixer that generates an excitation signal from the excitation component and a noise component, an inverse lattice filter that generates a modified residual echo from the excitation signal, and a gain adjuster that adjusts a power of the modified residual echo.

Abstract: The present invention is a computer method and apparatus of point slicing in digital communication systems employing Quadrature Amplitude Modulation (QAM). This method comprises the steps of providing an infinite lattice, receiving at least one subset of soft points, defining at least one valid constellation boundary, determining a closest valid constellation hard point for each soft point in the subsets of soft points, and reporting the closest valid constellation hard point.

Abstract: A signal classifier decomposes a subject signal into subbands. The signal classifier then analyzes the subbands for a presence of energy and uses detection logic to classify the energy as being representative of a frame of a signal using one of the following protocols: DTMF, MF-R1, ANS (V.25), LEC-DIS, V.21, or AA (i.e., telephone, facsimile, or modem). The detection logic is expandable to detect future protocols. The classifier may employ a preclassifier to select a reduced number of notch filters from among plural notch filters at frequencies which sinusoidal signals composing the energy in the subbands of the protocols may be found. A look-up table having predetermined ranges of data corresponding to the frequencies of the sinusoidal signals may be used to reduce calculation times. By splitting the subject signal into subbands and using efficient filters, the classifier uses an order of magnitude fewer processor instruction cycles than previous classifiers.