Abstract: A vector space comprising all of the filter coefficients for different interfering signals is provided, based on an acquired impulse response. A linear vector sub-space comprising all of the optimal filter coefficients for different interfering signals is established from said vector space using a vector space optimization method. The filter coefficients are established from the vector sub-space according to the current interfering signal being detected, during ongoing data transmission and at the maximum transmission speed.

Abstract: The partitioned block frequency domain adaptive filter according to the invention comprises a plurality of parallel arranged filter partitions. Each filter partition models a part of an impulse response of the adaptive filter and has update means for updating filter coefficients of that filter partition by means of a circular convolution. The update means intermittently constrain these filter coefficients by eliminating circular wrap-around artifacts of the circular convolution. The update means comprise selection means for selecting and removing at least part of the circular wrap-around artifacts. The selection means can be implemented with a relatively low computational complexity by means of an approximation of a rectangular constraint window.

Abstract: A system and method facilitating signal enhancement utilizing an adaptive filter is provided. The invention includes an adaptive filter that filters an input based upon a plurality of adaptive coefficients and modifies the adaptive coefficients based on a feedback output. A feedback component provides the feedback output based, at least in part, upon a non-linear function of the acoustic reverberation reduced output. Optionally, the system can further include a linear prediction (LP) analyzer and/or a LP synthesis filter. The system can enhance signal(s), for example, to improve the quality of speech that is acquired by a microphone by reducing reverberation. The system utilizes, at least in part, the principle that certain characteristics of reverberated speech are measurably different from corresponding characteristics of clean speech. The system can employ a filter technology (e.g. reverberation reducing) based on a non-linear function, for example, the kurtosis metric.

Abstract: An adaptive signal processing system utilizes a Multistage Weiner Filter having an analysis section and a synthesis section that includes a processor. The processor includes an algorithm for generating a data adaptive linear transformation, computing an adaptive weighting wmed of the data, and applying the computed adaptive weighting wmed to a function of a main input signal and an auxiliary input signal to generate an output signal. A plurality of building blocks in a Gram-Schmidt cascaded canceller-type configuration sequentially decorrelate input signals from each other to produce a single filtered output signal. Each building block generates an adaptive weight wmed that is applied to generate a local output signal. The effect of non-Gaussian noise contamination on the convergence MOE of the system is negligible. In addition, when desired signal components are included in weight training data they cause little loss of noise cancellation.

Abstract: The invention relates to a method of suppressing pulsed signals in particular of DME or TACAN type present in the radio signals received (Ue) by a radio-frequency receiver, characterized in that the reception frequency band of the receiver is divided into frequency sub-bands corresponding to the transmission channels of the pulsed signals, in that the presence of the pulsed signals and the transmission channel of said pulsed signals in the frequency sub-bands are detected, and in that the frequency sub-band comprising the detected pulsed signals is filtered over the duration of the pulsed signal so as to eliminate said pulsed signals pulse type.

Abstract: A system and method are disclosed for a nonlinear filter having a nonlinear transfer function. The nonlinear filter comprises a plurality of linear filters each having a filter output, a plurality of nonlinear elements each connected to one of the plurality of linear filters, and a combination network connected to the plurality of nonlinear elements. The nonlinear elements are used to produce nonlinear effects and generate a plurality of nonlinear outputs, and the combination network combines the nonlinear outputs.

Abstract: In some embodiments of the present invention, a known input signal is fed to a programmable filter. The output of the filter is compared to a calculated desired output for the same or similar input, and the difference between the outputs may be used to define correction values. The correction values are used then to correct an output signal.

Abstract: An apparatus and method for implementing an equalizer which (1) combines the benefits of a decision feedback equalizer (DFE) with a maximum-a-posterori (MAP) equalizer (or a maximum likelihood sequence estimator, MLSE) (2) performs equalization in a time-forward or time-reversed manner based on the channel being minimum-phase or maximum-phase to provide an equalization device with significantly lower complexity than a full-state MAP device, but which still provides improved performance over a conventional DFE. The equalizer architecture includes two DFE-like structures, followed by a MAP equalizer. The first DFE forms tentative symbol decisions. The second DFE is used thereafter to truncate the channel response to a desired memory of L1 symbols, which is less than the total delay spread of L symbols of the channel. The MAP equalizer operates over a channel with memory of L1 symbols (where L1<=L), and therefore the overall complexity of the equalizer is significantly reduced.

Abstract: An equalizer apparatus comprises a first filter (406) having a first respective plurality of filter coefficients, a channel model filter (414) having a plurality of model filter coefficients, and an adaptive algorithm unit (416). The adaptive algorithm unit (416) is arranged to adapt at least a first predetermined number of the first respective plurality of filter coefficients and at least a second predetermined number of the plurality of model filter coefficients in response to an error signal (e) corresponding to a difference in filter output signals from the first filter and the channel model filters (406, 414). The adaptive algorithm unit (416) operates in accordance with a respective state channel estimation technique. The respective state channel estimation technique is adapted so as to reduce a number of states allocatable to at least one of the plurality of state-defining taps, thereby reducing an overall number of states definable associated with the plurality of state-defining taps.

Abstract: An adaptive filter that, in one embodiment, filters rows of pixels of an image in a vertical direction, stores the results in row vectors, and then filters the row vectors in the horizontal direction, and displays the results or stores the results for later display. Coefficients of a reference filter are modified based on the output from the reference filter through a table-lookup process that accesses tables of modified filter coefficients. The output of the modified filter is added to a delayed version of the input to provide the adaptive filter output.

Abstract: A receiver includes a Viterbi-like equalizer that provides diversity combining of soft values to produce reliability information. The output reliability information at time k is the average of the first reliability information at time k and the second reliability information at time (k?1) after being normalized by the noise power. The first reliability information at time k is the difference between the two accumulated metrics of the two preceding nodes arriving at the same node having the global minimum node metric at time k over all transitions of all states. The second reliability function at time k is the difference between the best accumulated metric characterized by the last (L?1) bit being binary “one” and the best accumulated metric characterized by the last (L?1) bit being binary “zero.

Abstract: Provided is an adaptive filter that can reduce the computational complexity. The adaptive filter includes: a segmentation unit for segmenting N number of input signals into G number of signal groups; sub-filter unit having G number of sub-filters, which are corresponding to each of the signal groups, for filtering the corresponding signal group; an addition unit for summating the output signals of the sub-filter unit; an error computing unit for generating an error signal by comparing the output signals of the addition unit with a desired signal; filter coefficient updating unit having G number of filter coefficient updating units, each of which is corresponding to each of the sub-filters, for updating the filter coefficient of the corresponding sub-filter; and a switching unit for inputting the error signal to any one of the filter coefficient updaters optionally with respect to an iteration number k.

Abstract: Herein disclosed is a digital filter coefficient setting apparatus which can search and select a specified filter coefficient element corresponding to a specified coefficient parameter inputted by a parameter inputting unit from among the plurality of filter coefficient elements stored in a coefficient storing unit and calculate a specified filter coefficient element on the basis of the specified coefficient parameter when the specified filter coefficient element is not selected from among the plurality of filter coefficient elements stored in the coefficient storing unit, thereby enabling to efficiently set a digital filter in response to a specified filter coefficient element corresponding to any possible specified coefficient parameter inputted therein, as well as to reduce a time required for obtaining the specified filter coefficient element.

Abstract: Adaptive Finite Impulse Response Filter control includes structure and steps for receiving an input signal, filtering the input signal with an FIR filter having a plurality of filter stages, and delaying application of the input signal to at least one of said filter stages with respect to the other filter stages to skip filtering a portion of the input signal.

Abstract: A Finite Impulse Response (FIR) filter is provided including a coefficient generator to generate first and second coefficients, a first control conductor, and a second control conductor. A controller is coupled to a first end of the first control conductor and a first end of the second control conductor. A shared wiring is provided having its first end coupled to the coefficient generator. A first memory is coupled to a second end of the shared wiring and coupled to a second end of the first control conductor to store the first coefficient in response to the controller. A first multiplier is responsive to the first coefficient stored in the first memory and the input, and a first delay circuit is responsive to an input.

Abstract: A method for analyzing data, the data characterized by a set of scalars and a set of vectors, to analyze the data into components related by statistical correlations. In preferred embodiments, the invention includes steps or devices for, receiving a set of a scalars and a set of vectors as the inputs; calculating a correlation direction vector associated with the scalar and vector inputs; calculating the inner products of the input vectors with the correlation direction vector; multiplying the inner products onto the correlation direction vector to form a set of scaled correlation direction vectors; and subtracting the scaled correlation direction vectors from the input vectors to find the projections of the input vectors orthogonal to the correlation direction vector. The outputs are the set of scalar inner products and the set of vectors orthogonal to the correlation vector. The steps or devices can be repeated in cascade to form a multi-stage analysis of the data.

Abstract: An echo canceller includes an adaptive digital filter that generates an estimated echo signal {circumflex over (z)}[k] in response to (i) a sampled input data sequence x[k] and (ii) an error signal sequence e[k] indicative of the difference between a far end signal sequence y[k] and the estimated echo signal {circumflex over (z)}[k]. The adaptive filter includes N filter taps that each provide an associated tap output signal, wherein the adaptive digital filter generates the estimated echo signal {circumflex over (z)}[k] using the associated tap output signals from M of the N filter taps selected in response to a time delay estimate signal. The adaptive filter computes filter coefficients for each of the M number of the N filter taps using the associated tap output signals from the M number of said N filter taps.

Abstract: The partitioned block frequency domain adaptive filter according to the invention includes a plurality of parallel arranged filter partitions. Each filter partition models a part of an impulse response of the adaptive filter and has update unit for updating filter coefficients of that filter partition by a circular convolution. The update unit intermittently constrain these filter coefficients by eliminating circular wrap-around artifacts of the circular convolution. The update unit is arranged for updating the filter coefficients in dependence on at least part of the circular wrap-around artifacts, resulting in an improved convergence behavior of the adaptive filter.

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: In the MPEG2 Advanced Audio Coder (AAC) standard, Temporal Noise Shaping (TNS) is currently implemented by defining one filter for a given frequency band, and then switching to another filter for the adjacent frequency band when the signal structure in the adjacent band is different than the one in the previous band. The AAC standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. This current practice is not an effective way of deploying TNS filters for most audio signals. We propose two solutions to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique.

Abstract: A data communication receiver comprises an equalizer for adapting to each of a plurality of channels to open the eye for each channel in a Gigabit (1000BASE-T) transceiver. The eye is open for a first channel (A) and a transformation process applies the coefficients of that adaptation to open the eye for the other dimensions. The transformation process keeps the magnitude response constant.

Abstract: A method of conditioning a signal being communicated between a system to be controlled and a controller may include monitoring an actual output signal and conditioning the actual output signal. The method also includes determining the difference between the actual output signal and the condition signal and causing the actual output signal to be filtered based on the relationship between the difference between the actual output signal and the conditioned signal.

Abstract: A circuit (48) and method, which can be used in a mass data storage device, controls adaptation asymmetry of coefficients of an FIR filter (20) using an accumulator (52) or accumulating correlation results between unequalized FIR filter input data samples and FIR filter output equalized error samples. A circuit (52) generates coefficient increment and decrement requests from the accumulated correlation results. A circuit (120,102?,122) updates the coefficients within a symmetric coefficient pair on the basis of the increment and decrement requests only if a predetermined nonzero coefficient magnitude difference between the coefficient pair would not be exceeded by the update.

Abstract: The present invention provides an adaptive filter. In one embodiment, the adaptive filter includes a solution vector generator that develops a sparse expression of an initial solution vector. In addition, the adaptive filter includes a proportionate normalized least mean squares (PNLMS) analyzer, coupled to the solution vector generator, that employs the sparse expression to converge upon at least one coefficient for the adaptive filter.

Abstract: In order to transform an input digital signal (xn) into one or more output digital signals (yn) containing even-indexed samples (y2n) and odd-indexed samples (y2i+n), this filtering method includes at least one iteration which contains an operation of modifying even-indexed samples (y2n) by a function (R) of weighted odd-indexed samples (y2n+1), and an operation of modifying odd-indexed samples (y2n+1) by a function (R) of weighted even-indexed samples (?0,j(y2n?y2n+2)). The weighted samples are obtained by at least one weighting operation. At least one of the weighting operations is applied to the difference between two consecutive even-indexed samples.

Abstract: Stereo echo cancellation is necessary to overcome the objections observed by, for example, teleconferencing, voice controlled video/audio apparatuses, etc. To improve the existing filters, an adaptive filter is used along with a signal processing device which obtain the coefficient updates in the transformed domain, reducing the required calculation complexity. Further, the filter includes circuitry for reducing the correlation between the input signals on the coefficient updates.

Abstract: A signal filtering apparatus and associated methods enable noise to be significantly reduced or eliminated from a signal. In a described embodiment, the signal is indicative of tension in a slickline. An adaptive filter is used to effectively cancel the noise from the signal, using an input signal characteristic of a noise source.

Abstract: The present invention relates to a filter bank approach to adaptive filtering method using independent component analysis. More particularly, the invention relates to a method of improving the performance of adaptive filtering method by applying independent component analysis that is capable of reflecting the secondary or even higher order statistical characteristics to adaptive filtering algorithm using the filter bank approach. In order to implement the conventional adaptive filter algorithm using independent component analysis to the real world problem, a large number of filter training coefficients are required and also a large amount of calculation is required when a training is undertaken. This results in a very slow learning speed and the deterioration in the quality of result signals.

Abstract: An FIR filter and setting method of filter coefficients thereof enabling the prevention of breakdown of equi-ripple of weighted approximation error and preservation of gain of the pass band into approximately constant. The setting method includes five steps. The initial setting step is that setting of the FIR filter, setting of the band, setting of coefficients of a pre-filter, and setting of initial extreme value point. The first step is that interpolation polynomial equation is generated for interpolating amplitude characteristic from the present extreme value point of the amplitude characteristic of the frequency. The second step is that a new extreme value point is determined from the amplitude characteristic obtained from the interpolation polynomial equation that is obtained in the first step. The third step is that judgement is performed whether or not position of the extreme value is approximated within required range while repeating the first step and the second step.

Abstract: Optimal Decision Feedback Equalizer (DFE) coefficients are determined from a channel estimate h by casting the DFE coefficient problem as a standard recursive least squares (RLS) problem, e.g., the Kalman gain solution to the RLS problem. A fast recursive method, e.g., fast transversal filter (FTF) technique, for computing the Kalman gain is then directly used to compute Feed Forward Equalizer (FFE) coefficients gopt. The complexity of a conventional FTF algorithm is reduced to one third of its original complexity by choosing the length of a Feed Back Equalizer (FBE) coefficients bopt (of the DFE) to force the FTF algorithm to use a lower triangular matrix. The FBE coefficients bopt are then computed by convolving the FFE coefficients gopt with the channel impulse response h. In performing this operation, a convolution matrix that characterizes the channel impulse response h extended to a bigger circulant matrix.

Abstract: A dual-line DSL system comprises a central office and a customer premises coupled by at least two communication paths that together provide a single high-bandwidth channel. An ADSL transceiver unit-remote (ATU-R) at the customer premises is configured to reduce the effects of far end cross talk and near end cross talk on signals received by the ATU-R.

Abstract: An apparatus and method for implementing an equalizer which combines the benefits of a decision feedback equalizer (DFE) with a maximum-a-posterori (MAP) equalizer (or a maximum likelihood sequence estimator, MLSE) to provide an equalization device with significantly lower complexity than a full-state MAP device, but which still provides improved performance over a conventional DFE. The equalizer architecture includes two DFE-like structures, followed by a MAP equalizer. The first DFE forms tentative symbol decisions. The second DFE is used thereafter to truncate the channel response to a desired memory of L1 symbols, which is less than the total delay spread of L symbols of the channel. The MAP equalizer operates over a channel with memory of L1 symbols (where L1<=L), and therefore the overall complexity of the equalizer is significantly reduced.

Abstract: An equalizer for use in a communication receiver includes an infinite impulse response (IIR) feedback filter operated in acquisition and tracking feedback modes on a sample by sample basis to form a hybrid Decision Feedback Equalizer (DFE) architecture. In acquisition mode, soft decision samples from the filtered received signal are input to the IIR filter. In the tracking mode, hard decision samples from a slicer are input to the IIR filter. Acquisition and tracking operating modes are selected in accordance with a set of decision rules on a sample by sample basis based on the quality of the current hard decision. If the current hard decision is low quality, then the soft decision sample (acquisition mode) is used. If the current hard decision is high quality, then the hard decision sample (tracking mode) is used. In such manner, the DFE is operated in a hybrid mode, i.e., using both soft and hard decisions on a sample by sample basis.

Abstract: A method for processing a signal propagated through a nonlinear channel is disclosed. The method comprises modeling the channel characteristics to produce a linearized channel model, wherein the linearized channel model has a linearized transfer function that includes a plurality of first order polynomials and nonlinear operators. The method further comprises deriving an inverse linearized channel model from the linearized channel model and filtering the signal using the inverse linearized channel model.

Abstract: A signal-processing circuit in which two-stage equalization is performed by using first and second equalization circuits provided on upstream and downstream sides from a phase-locked loop circuit, respectively, the first equalization circuit on the upstream side including a transversal filter to minimize an equalization error caused by the first equalization circuit and to stabilize the operation of the phase-locked loop circuit. Another signal-processing circuit including an analog-to-digital converter and a digital phase-locked loop circuit for receiving the output from the analog-to-digital converter and a recording and playback apparatus using the output are also provided, wherein the output from the analog-to-digital converter is input as the digital signal in the digital phase-locked loop circuit to fetch a detection point voltage and to stabilize the phase-locked loop circuit without an analog circuit.

Abstract: A method for deriving a polynomial for generating filter coefficients to be used in a digital filter in an electronic unit having a digital signal processor. The digital filter has a plurality of reconfigurable filter sections to achieve a desired bandwidth characteristic between a first characteristic with a first cutoff frequency and a second characteristic with a second cutoff frequency. A first set of filter coefficients is determined for providing the first characteristic. A second set of filter coefficients is determined for providing the second characteristic. A third set of filter coefficients is determined for providing a third characteristic having a third cutoff frequency between the first and second cutoff frequencies.

Abstract: A variable-gain digital filter includes a selector and multiplier arranged inside the digital filter for regulating gain whereby the number of bits of filter input is X, the number of flip-flops inside the filter is X×n bits, and a (Y×n bit) reduction in the number of flip-flops is enabled. The gain regulation circuit incorporated within the digital filter enables a reduction in circuit scale.

Abstract: A method and apparatus for adaptively removing interference from a signal. In one embodiment, the invention comprises an infinite impulse response (IIR) notch filter configured to receive a wideband signal and provide a filtered output signal, and a controller coupled to the notch filter to adaptively control the null frequency of the notch filter thereby removing narrowband interference from the received wideband signal. The controller may employ a gradient-based algorithm to detect the highest power frequency band in the output signal and modify the null frequency of the notch filter to minimize the power of the output signal.

Abstract: A variable sample rate recursive digital filter is an adaptive digital filter where its coefficients are recalculated for each sample rate being processed in such a way as to maintain a constant frequency rate for all sample rates. An equivalent resampling is done by taking the ratio of the bilinear transforms at the respective sample rates. From an initial or calibrated sample rate and a corresponding initial filter coefficient, a new filter coefficient for a new sample rate is obtained by multiplying the initial filter coefficient by a constant or coefficient factor that is a function of the initial filter coefficient and a ratio of the initial and new sample rates: zFactor(z,R):=(1/z){(z(1+R)+(1?R))/(z(1?R)+(1+R))} The resulting new filter coefficient provides the adaptive digital filter with a constant frequency response when compared to the initial sample rate frequency response.

Abstract: A filter adaptation unit for producing a set of filter coefficients indicative of a transfer function of a system in a given state is provided including a coefficient set selection unit for selecting between two or more sets of filter coefficients. For each set of filter coefficients a respective set of error characterization data elements is generated characterizing the error in a filter's impulse response when using the corresponding set of filter coefficients. The selection unit provides functionality for processing the first and second sets of filter coefficients and the second set of error characterization data elements in order to detect whether a change in the state of the system has occurred and to select a preferred set of filter coefficients. The selected set of filter coefficients is released in a format suitable for use by an adaptive filter.

Abstract: An iterative method of equalizing an input signal received over a digital communication channel can include (a) using a kernel density estimate where different values of a kernel size are indicative of either a blind or a decision-directed equalization mode, (b) processing a received signal using a blind equalization mode, and (c) evaluating, on a block or sample basis, an error measure based on a distance among a distribution of an equalizer output and a constellation. The method also can include (d) updating the kernel size based upon the error measure thereby facilitating automatic switching between the blind and decision-directed equalization modes, where the kernel size is initially set to a value indicative of the blind equalization mode. The method additionally can include (e) selectively applying blind equalization or decision-directed equalization to the input signal according to the updated kernel size for subsequent iterations of steps (c)-(e).

Abstract: A control apparatus includes a manipulated variable output unit, calculation unit, first lower limit setting unit, first upper limit setting unit, second lower limit setting unit, second upper limit setting unit, and controlling element. The manipulated variable output unit outputs first and second manipulated variables to an object to be controlled. The calculation unit calculates a limit cycle auto-tuning control parameter. The controlling element performs feedback control calculation based on the control parameter for the deviation between a set point and a controlled variable to calculate the first manipulated variable, and outputs the calculated first manipulated variable to the object.

Abstract: A filter adaptation unit suitable for producing a set of filter coefficients is provided including an error characterization unit for characterizing the error in a filter's impulse response. The error characterization unit generates a set of error characterization data elements associated to a newly generated set of filter coefficents. A selection unit then makes a selection between one of the newly generated set of filter coefficients and an existing set of filter coefficients at least in part on the basis of their respective sets of error characterization data elements. The selected set of filter coefficients is then released in a format suitable for use by an adaptive filter.

Abstract: An improved adaptive line enhancer includes an adaptive Gray-Markel lattice notch filter having an adaptive notch frequency, in which the notch frequency is determined according to a notch frequency variable k. The value of k for the n+1th sample period is determined according to the following equation: k(n+1)=k(n)?sgn[y(n)]sgn[UPDATEFN]×? in which y(n) is the notch filter output, ? is the adaptation constant, and UPDATEFN has a transfer function in the z-transform domain of: ( ? - 1 ) ? ( k ? ( n ) - 1 ) ? z - 1 1 + k ? ( n ) ? ( 1 + ? ) ? z - 1 + ? ? ? ? z - 2 in which ? determines the bandwidth and k(n) is a variable for determining the current notch frequency. The algorithm for adapting the notch frequency enables the notch frequency to be directly calculated from knowledge of internal variables of the wave digital filter.

Abstract: The present invention uses narrow-band monobit receivers after a digital filter bank to separate simultaneous signals in one channel. The invention improves the capability of wideband digital receivers.

Abstract: The present invention is directed to an echo canceller adapted for use in a communication system that includes a hybrid circuit. The echo canceller comprises an adaptive digital filter that generates an estimated echo signal {circumflex over (z)}[k] in response to: (i) a sampled input data sequence x[k] and (ii) an error signal sequence e[k] indicative of the difference between a near end signal sequence y[k] and the estimated echo signal {circumflex over (z)}[k]. The adaptive digital filter computes filter coefficients based upon the error signal sequence e[k] using a stochastic quadratic descent estimator, such as for example a least mean square (LMS) estimator, that employs a dynamically adjustable step size vector ?[k].

Abstract: For the offset compensation of a digital signal, particularly of a communication signal transmitted in a cordless digital communication system, a recursive digital filter is used. The recursive digital filter has at least one filter coefficient that is varied in a time-dependent manner. The recursive digitial filter has a first multiplying device multiplying symbols of the digital input signal by a first time-variable filter coefficient to obtain a digital output signal having symbols.

Abstract: An adaptive signal processing system utilizes a pseudo-median cascaded canceller to compute a set of complex adaptive weights and generate a filtered output signal. The system includes a plurality of building blocks arranged in a Gram-Schmidt cascaded canceller-type configuration for sequentially decorrelating input signals from each other to thereby yield a single filtered output signal. Each building block includes a local main input channel which receives a local main input signal, a local auxiliary input channel which receives a local auxiliary input signal, and a local output channel which sends a local filtered output signal. Each building block generates a complex adaptive weight which is the sample median value of the real and imaginary parts of the ratio of local main input weight training data to local auxiliary input weight training data, and each building block generates a local output signal utilizing the complex adaptive weight.

Abstract: An apparatus and method for generating finite impulse response (FIR) filter coefficients are presented. The apparatus includes an address generator that multiplies a desired cutoff frequency f by an integer n to generate an address, a first look-up table that generates a sine function value of the address, a divider that divides the sine function value by n*pi, a multiplexer that generates an impulse response function value by selecting one of a value produced from the divider and 2*f based on an outside control signal, and a multiplier that multiplies the impulse response function value by a corresponding window function value to generate an nth filter coefficient for the FIR filter.

Abstract: A method is provided for reducing power dissipation within a communications system having a plurality of adaptive filters with a plurality of taps, each tap is switchable between an active and an inactive state, each tap also has a coefficient. An acceptable error for the system is specified. This error is typically the mean squared error of the system. A tap threshold is set for each active tap. Those taps having a coefficient with an absolute value less than the tap threshold set for the active tap are deactivated. The error of the system is computed and compared to the acceptable system error. If the computed system error is less than the acceptable system error, the tap threshold for each active tap is increased.