Abstract: The device receives an input signal xt and an observation signal yt, one component of which is the response of a system to the input signal. It determines an error signal et at an instant t in accordance with the equation et=yt?XtTHt?1, where Ht?1 is a vector made up of L coefficients of an identification filter having a finite impulse response representative of the impulse response of the system and XtT=(xt, xt?1, . . . , xt?L+1). A predictive analysis of the input signal is performed by a computation applied to frames of the input signal. The L coefficients of the identification filter are adapted by adding to the vector Ht?1 a vector proportional to e t X t T ? U t + ? ? U t , where Ut is a vector made up of the L values of a prediction residue of the input signal at the instant t and at the preceding L?1 instants, and ? is a positive or zero coefficient.

Abstract: Parallel adaptive filters and filtering methods that enable processing of an input signal in a circuit that has an clock speed many times slower than the input rate of the input signal that is processed. A polyphase decimator structure processes a data stream requiring a low pass filtered bandlimited (low-rate) output that is used for high-rate output structures. The filters and methods break an input data stream into parallel paths that efficiently produce a bandlimited (decimated, low-rate) filtered output. Each of the parallel paths is processed at a decimated rate to provide a filtered output signals corresponding to a filtered version of the input signal.

Abstract: An echo cancellation invention which does away with the need for an accurate double-talk detector, while maintaining a higher adaptation gain for quicker convergence and also providing increased stability. It does this by operating two filter models of the acoustic path in parallel. One model is adapted continuously to find the most accurate model of the echo path, while the other is not. The echo canceller output is taken from the filter that is not adapted. A comparison of the residual error (echo) is done between the model being adapted and the model being listened to at regular intervals. When the model being adapted has less error (echo) than the model being listened to its filter coefficients are copied to the other model. If the model being adapted has greater error (echo) than the other model (caused by noise diverging the adaptation process) then the adapted model has its coefficients overwritten by the filter that is listened to.

Abstract: In one embodiment of the invention, a tap-leakage generator includes an error filter and an updater. The error filter filters a decision error provided by the adaptive filter using a leakage factor. The adaptive filter has N taps. The updater updates N equalizer coefficients to the N taps using the filtered decision error. The updater receives N equalizer data from the N taps.

Abstract: Provided is a normalizing apparatus for adaptive beamforming by performing a normalizing process which uses a normalized least mean square (NLMS) algorithm that produces a weight vector for adaptive beamforming, in a smart antenna receiver. For the normalizing process, the normalizing apparatus includes a multiplication operation means that performs a multiplication operation, and a division operation means that performs a division operation using mathematic calculations based on binary logarithm principles, and using addition and subtraction operations.

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: Presented herein is a system and apparatus for accelerating arithmetic decoding of encoded data. In one embodiment, there is presented a symbol interpreter for decoding CABAC coded data. The symbol interpreter comprises a first memory, a CABAC decoding loop, and a syntax assembler. The first memory receives a bitstream comprising the CABAC coded data at a channel rate. The CABAC decoding loop decodes the CABAC symbols at the channel rate, and comprises an arithmetic decoder for generating binary symbols from the CABAC coded data at the channel rate. The syntax assembler decodes the binary symbols at a consumption rate.

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: An active noise control system is provided which cancels a noise using a sound radiated from a speaker driven by an output from an adaptive notch filter. The system employs output signals from an adder or simulation cosine-wave and sine-wave signals, an error signal or an output signal from a microphone, and a compensated signal from the adder or a signal available for acoustically transferring an output from the adaptive notch filter to the microphone in accordance with initial transfer characteristics to update the filter coefficient of the adaptive notch filter. This configuration allows the system to operate with stability even when the acoustic transfer characteristics vary with time or under circumstances where there exists a significant amount of incoming external noises. The system also prevents overcompensation for a noise at the ears of a passenger in a vehicle, thereby proving an ideal noise reduction effect.

Abstract: Method and apparatus for constraining tap coefficients in an adaptive Finite Impulse Response filter includes structure and steps whereby a coefficient supply circuit provides at least two even tap coefficients and at least two odd tap coefficients to the adaptive Finite Impulse Response filter. Constraint circuitry then selectively constrains changes in the values of at least one of (i) the two even tap coefficients and (ii) the two odd tap coefficients. Preferably, the Finite Impulse Response filter has taps C0, C1, C2, C3, C4, C5, and C6. A coefficient supplier is coupled to provide coefficients to said taps, and an adaptive circuit changes the coefficients supplied by said coefficient supplier in accordance with changes in an output of the Finite Impulse Response filter. The adaptive circuit includes circuitry to constrain allowable change in both the even tap coefficients C0, C2, C4, C6 and in the odd tap coefficients C1, C3, C5.

Abstract: A flexible adaptation engine includes a coefficient adaptation circuit that implements multiple adaptation algorithms, and/or multiple coefficient selection algorithms, to adapt the filter coefficients of one or more digital filters, such as the transversal filters of a receiver. In one embodiment, a controller selects the filter coefficients to be adapted, and the adaptation algorithm(s) to be used to adapt the selected coefficients, based on various criteria such as convergence status data, clock recovery status signals, the current load on a processor that adapts the coefficients, and/or manual control signals. In one embodiment, the architecture supports the ability to vary the number of coefficients that are updated at a time, and to concurrently apply different adaptation algorithms to different subsets of filter coefficients. The flexible adaptation engine may be implemented in application-specific hardware and/or as a processor that executes software.

Abstract: A filter engine that performs filtering operations on an input data stream comprising blocks of data. The filter engine includes a first memory element, a second memory element, a first shift register, a second shift register and a processor. The first and second memory elements store blocks of data to be processed. The first shift register receives and stores blocks of data from the first memory element. The second shift register receives and stores blocks of data from the second memory element. The first and second shift registers are adapted to selectively shift their contents by a predetermined number of bits corresponding to the size of a data element, such as a pixel. The processor receives blocks of data from the first and second shift registers and simultaneously performs filtering operations on blocks of data from the first and second shift registers.

Abstract: An improved method of high pass filtering a data set includes flattening the data set and then filtering the flattened data set with an adaptive filter. The data set is flattened by fitting it to a predetermined function, and then obtaining the difference between the original data set and the fitted data set. Beneficially, the predetermined function is a polynomial. The adaptive filter includes a masking function that has a constant, non-zero value (e.g., 1) within the bounds of the original data set and value of zero outside the bounds of the original data set.

Abstract: A Fast Affine Projection (FAP) adaptive filter and method of adaptive filtering are disclosed, which reduce instability associated with FAP filters caused by error accumulation in the process of inversion of an autocorrelation matrix. The method provides updating of the adaptive filter coefficients by solving at least one system of linear equations whose coefficients are the autocorrelation matrix coefficients, by using a descending iterative method with intrinsic feedback. The results of the solution are used to update the adaptive filter coefficients. The approach is applicable for a normalized step size ranging from zero to unity, and allows either direct determination of updated filter coefficients without determining an inverse autocorrelation matrix, or, determining the inverse autocorrelation matrix by a descending iterative method. In some embodiments, a normalized step size is set close to unity, and the system of linear equations is solved by steepest descent or conjugate gradients methods.

Abstract: Multiported register files used for storing coefficients in adaptive FIR are improved upon by implementing a split memory architecture that has the ability to separately control the least significant bits and the most significant bits of coefficient values that are stored in the filter. When the filter is operated to use so-called “burst mode” updating, the updating circuitry of the filter can be disabled and only the most significant bits of the coefficients are read out from the multiported register file while the least significant bits remain unchanged. This conserves power without sacrificing precision, since only certain ones of the bits of the coefficients are used in the multiplication of the sample.

Abstract: A reflection sound generator has a first filter of Finite Impulse Response (FIR) type that is provided with a first set of parameters representing a first distribution pattern of reflection sounds, and a second FIR-type filter provided with a second set of parameters representing a second distribution pattern of additional reflection sounds. The first distribution pattern has a time length sufficient to cover an initial reflection sound and subsequent reverberant reflection sounds which are distributed at intervals along the time. The first filter executes convolution operation of sample data of an input sound by the first set of parameters to generate first data containing a sequence of reflection sounds of the input sound. The second filter executes convolution operation of the first data by the second set of parameters to generate second data containing additional reflection sounds which fill the intervals of the reflection sounds of the input sound.

Abstract: The present invention proposes a device for processing received signals (y) having been transmitted via a transmission channel (1), comprising: estimation means (2, 2a) adapted to obtain an estimated impulse response function (hk) of said channel (1) based on said received signals (y) which are received at a first time; derivation means (3; 3a-3f) adapted to derive history information (wink) based on at least one previously estimated impulse response function, calculation means (4; 4a, 4b) adapted to calculate modifying information (mwink) on the basis of said history information (wink), and modification means (5) adapted to modify said estimated impulse response function (hk) of said channel (1) obtained on the basis of said received signals (y), by applying said modifying information (mwink) to said estimated impulse response function (hk) of said channel obtained on the basis of said received signals (y), and adapted to output a modified impulse response function (hmk).

Abstract: A novel method for implementing a filter for processing discrete signal image is presented having the steps of first obtaining a plurality of sample values from said discrete signal image and then using each sample value to retrieve a bit vector from a plurality of tables. Afterwhich a logical AND on the set of retrieved bit vectors is performed and the position of the largest non-zero bit is determined. Then, this position is used to index into a table of filter values. The corresponding filter value is then retrieved from the table.

Abstract: A configurable adaptive filter that is used for echo cancellation is disclosed, which includes a method of detecting a voice or no-voice signal. The presence of a voice or no-voice signal is determined by calculating a histogram of signal amplitude value over a period of time. If this histogram has more than a predefined number of samples that are above a threshold then the signal is classified as no-voice or periodic otherwise the signal is classified as a voice signal. A variable maximum amplitude limit and lower amplitude thresholds are disclosed to detect a voice or no-voice from the histogram signal faster than traditional methods utilized in echo cancelers. A configurable hysteresis time is used to ensure the signal register primarily contains voice signal when the filter coefficients of the echo canceler are allowed to adapt.

Abstract: Methods and systems for echo cancellation are provided. According to one method, when a far end signal with an echo having a major body and a pure delay portion is received, a filter having a finite length is provided. Once the finite length filter covers a portion of an echo path, the method includes determining whether at least a portion of the echo major body is within the filter. In one embodiment, the filter is divided into a plurality of sub-windows, and the step of determining whether the filter covers the major body of the echo path includes using filter coefficients of the first sub-window. According to one embodiment, the major body of the echo path is detected by comparing a maximum value coefficient in the first sub-window with a predetermined threshold value. If the maximum value coefficient is greater than the threshold value, the major body is detected.

Abstract: A method of echo cancellation for a signal transmission system in which the size of the step used to adapt filter coefficients is adjusted in accordance with the echo delay and in which stationary signals are avoided by determining when the mean and variance of coefficients obtained from a second order linear predictive coding analysis of successive far end samples exceed preset thresholds.

Abstract: A method for filtering signals from a nonlinear dynamical system is provided. An initial enhanced point is set to a noisy point obtained from the signals. An intermediate enhanced point is estimated. A new enhanced point is calculated using the estimated point and a weighting constant. Estimation of an intermediate enhanced point and calculation of a new enhanced point are iterated until the computed point converges to a true enhanced point which represents a noise reduced signal to provide filtered signals, wherein the new enhanced point is computed by the equation {circumflex over (x)}n(i)={circumflex over (x)}n(i−1)+K3[{circumflex over (x)}n(i,temp)−{circumflex over (x)}n(i−1)] wherein {circumflex over (x)}n(i) is the new enhanced point, {circumflex over (x)}n(i-temp) is the intermediate enhanced point for iteration i, {circumflex over (x)}n(i−1) is the enhanced point at iteration i−1, and K3 is a weighing constant.

Abstract: A digital data system (100) provides 1-D, 2-D and 3-D capability and multi-band channel capability. Improved filter banks are created by generating a filter bank having an analysis portion and synthesis portion and obtaining wavelet coefficients (302) for each portion. The wavelet coefficients are expressed in a format capable of canonical signed digit (CSD) representation, such as integers (302). The canonical signed digit (CSD) representation is controlled by a value, N, selected to control resolution of the CSD coding. Optimized CSD-coded wavelet coefficients are used as filters for data signals (318).

Abstract: The invention relates to a method for calculating a least mean square algorithm using an N-tap filter. A modified least mean square algorithm is used for the calculation. In a first calculation step, calculation is effected using a first, wide bit width in the N taps. In a second step, depending on the result of the first calculation step, a second, smaller number of bits is selected for the further method steps.

Abstract: Stable adaptive filter and method are disclosed. The invention solves a problem of instability associated with Fast Affine Projection adaptive filters caused by error accumulation in an inversion process of an auto-correlation matrix. The Stable FAP uses a simplification of setting a normalized step size close to unity and reduces a problem of the matrix inversion to solving a system of linear equations whose solution coincides with a first column of the inverse auto-correlation matrix. The system of linear equations is then solved by one of descending iterative methods which provide inherent stability of operation due to intrinsic feedback adjustment. As a result, inevitable numerical errors are not accumulated, being corrected as the process goes on. In first and second embodiments of the invention the system of linear equation is solved by steepest descent and conjugate gradient methods respectively. Being immune to numerical errors, the invented method and filter are suitable for various DSP platforms, e.

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: An adaptive filter suitable for network echo cancellation and other applications contains a coefficient vector update device for feeding coefficient vector updates to a finite impulse response filter in accordance with fast converging algorithms. A double talk detector is included for causing filter adaptation to cease in the presence of double talk in the system being echo cancelled. The coefficient vector update device utilizes a proportional affine projection algorithm to provide fast convergence of the filter system and improved performance over other filter devices utilizing different fast converging algorithms.

Abstract: An adaptive digital filter of the present invention includes: a pipelined filtering section for performing a filtering operation based on input data and coefficient data so as to output filtered data; and a non-pipelined adaptation section for outputting the coefficient data to the pipelined filtering section by performing a coefficient adaptation operation in a non-pipelined process based on the input data and the filtered data so that the filtered data output from the pipelined filtering section converges to a predetermined reference value.

Abstract: A method for dynamically regulating the power consumption of a high-speed integrated circuit which includes a multiplicity of processing blocks. A first metric and a second metric, which are respectively related to a first performance parameter and a second performance parameter of the integrated circuit, are defined. The first metric is set at a pre-defined value. Selected blocks of the multiplicity of processing blocks are disabled in accordance with a set of pre-determined patterns. The second metric is evaluated, while the disabling operation is being performed, to generate a range of values of the second metric. Each of the values corresponds to the pre-defined value of the first metric. A most desirable value of the second metric is determined from the range of values and is matched to a corresponding pre-determined pattern. The integrated circuit is subsequently operated with selected processing blocks disabled in accordance with the matching pre-determined pattern.

Abstract: The filtering coefficients of a frequency domain stereophonic echo canceller are adapted by a method which takes account of the cross-correlation between the input signals relating to the two channels. In particular, the adaptation process takes account of the coherence function, reducing the problems commonly encountered with stereophonic cancellation schemes when relatively correlated input signals occur.

Abstract: An adaptive filter allowing rapid adaptation to a sudden and great change in a system to be estimated is disclosed A memory stores position information indicative of a signal block connected to each of M filter circuits and power information obtained from K filter coefficients generated by each of the M filter circuits. A valid/invalid block selector selects one as an invalid block from the M signal blocks connected to the M filter circuits and determines position information indicative of a selected signal block as an invalid block and a corresponding filter circuit that has been connected to the selected signal block. The position information indicative of the selected signal block is appended to a queue of a shift register and position information is output from a head of the queue. Based on the corresponding filter circuit and the position information received from the shift register, a matrix switch connecting M filter circuits and M signal blocks is controlled.

Abstract: The rate of convergence of an adaptive system is optimized as a function of the stability with which it converges to a desired state. In the adaptive technique according to the invention, for optimal convergence, a large adaptive coefficient value is used in stable regions with relatively smaller noise, while a smaller adaptive coefficient is used in unstable or noisy environments. Also when the solution has been reached &mgr; is decreased to decrease the effects of noise and the residual error. Various algorithms are used to evaluate the stability with which the system converges and to select a corresponding adaptive coefficient. The method and apparatus according to the invention provides an optimum rate of convergence based on the overall stability of the system, while minimizing the residual error.

Abstract: A reconfigurable filter node including an input data memory adapted to store a plurality of input data values, a filter coefficient memory adapted to store a plurality of filter coefficient values, and a plurality of computational units adapted to simultaneously compute filter data values. Filter data values are the outputs of a filter in response to input data values or a second plurality of filter coefficients to be used in subsequent filter data value computations. First and second input data registers load successive input data values input data memory or from adjacent computational units. Each computational unit comprises a pre-adder adapted to output either the sum two input data values stored in the computational unit or alternately to output a single input data value, and a multiply-and-accumulate unit adapted to multiply the output of the pre-adder by a filter coefficient and accumulate the result.

Abstract: A method of conjoint detection of a set of CDMA codes received at a plurality of antennas of a receiver like a mobile telephone receiver. The codes are transmitted via a transmission channel with transfer matrix A satisfying the equation e=A.d+n, where d is the set of symbols of the codes transmitted, n is an additional noise vector and e is the set of received samples.

Abstract: An analog filter is connected at the output of a D/A converter. The analog filter eliminates higher frequency components than the original basic frequencies for a digital audio system and improves the phase characteristic. The analog filter comprises a plurality of band elimination filters that have respective cutoff frequencies of integral multiples of the sampling frequency fs. An audio amplifying circuit is connected to the analog filter. Each band elimination filter eliminates side band components close to integral multiples of the sampling frequency. By such construction, the analog filter has the necessary amplitude characteristic and lessens phase shift within the basic frequency.

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: A system and method for accelerating least-mean-square algorithm-based coefficient adaptation which executes in one machine clock cycle one tap of the least-mean-square algorithm including data fetch, coefficient fetch, coefficient adaptation, convolution, and write-back of a new coefficient vector. A data memory stores an input signal. A coefficient memory stores a coefficient vector. A multiplication and accumulation unit reads the input signal from the data memory and the coefficient vector from the coefficient memory to perform convolution.

Abstract: A method of providing real time digital pulse shaping includes: receiving a digital pulse input signal (31); applying the digital pulse input signal to first (33, 34, 35) and second (32, 36, 37) processing channels, the first processing channel including a CONCAVE shaper (34) and the second processing channel including a CONVEX shaper (36); applying selected digital control parameters to the CONCAVE shaper (34) and the CONVEX shaper (36) to produce desired first and second weigthing functions, and superposing the first and second weighting functions to produce a desired overall weighting function.

Abstract: The present invention is a system and system for equalizing a signal in a communication system. The signal is represented by samples at a time instant, and the samples are filtered using a sign permutation filter. An estimate of the signal is determined by a linear combination of the samples with corresponding weights. The error of the estimate is then computed. The weights are updated by an updating circuit to minimize the error.

Abstract: Operation of an adaptive filter includes determining a first filter coefficient and determining a second filter coefficient based on the first filter coefficient, in one embodiment, and in another embodiment includes comparing pseudo-error counts generated for each of several time intervals and setting filter coefficients based on the comparison. Adaptive filter coefficient generating apparatus includes a pseudo-error count generator adapted to receive the output of the adaptive filter and to generate counts of pseudo-errors occurring during successive time intervals, and a filter coefficient generator generating filter coefficients for the adaptive filter in response to the pseudo-error counts.

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: An adaptive filter having a hybrid structure and capable of operating in a tri-mode regime is disclosed. The tri-mode adaptive filter includes three sub-filters which operate in a fast convergence mode, a slow convergence mode, and a freezing mode respectively. The adaptive filter further comprises a quality detector which measures non-convergence and a near-end signal of the tri-mode filter and generates a feedback signal which is sent to a switching means. In response to the feedback signal, the switching means provides switching between the sub-filters so that one of the sub-filters operates at a time, providing adaptive filtering in accordance with its corresponding filtering method.

Abstract: An adaptive filter, a control method of the adaptive filter, and a storage medium storing therein a program for executing the control method thereof enable a tap weight control method to be provided in order to realize high speed convergence of an adaptive filter using sign-sign algorithm. The adaptive filter having a non-recursive filter obtains a correlation value in such a way that a sum between an error signal and an additive noise is multiplied by a value in every respective taps of input signal. The correlation value undergoes operation of a correlation non-linear processor to obtain a value. Such the value is multiplied by a step gain. An obtained product is utilized for updating above described respective tap weight coefficients. It causes a power-number of a power function to be controlled according to an estimated value of an electric power of the error signal while taking the non-linear function of the correlation non-linear processor to be the power function of the correlation value.

Abstract: A method for filtering a digital signal sequence with adaptive filters given assistance of discrete parameter wavelet transformation is recited that, among other things, can be utilized in telecommunication technology for echo compensation. The method has especially small filter lengths, and thus low calculating outlay, for determining the optimum coefficient set for the filters employed.

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: An online Gaussian mixture learning model for dynamic data utilizes an adaptive learning rate schedule to achieve fast convergence while maintaining adaptability of the model after convergence. Experimental results show an unexpectedly dramatic improvement in modeling accuracy using an adaptive learning schedule.

Abstract: In an adaptive type signal estimator, an estimation signal storage unit outputs the first sample signal to which nonlinear distortion is added. A convolution arithmetic unit adds linear distortion to the first sample signal and outputs the second sample signal. A coefficient corrector receives the first sample signal, reception signal, and the determination signal, and corrects the second sample signal. The coefficient corrector obtains a difference signal based on a replica generated from the reception signal and first sample signal and a delayed reception signal obtained by delaying the reception signal by a predetermined period of time, obtains a product signal by multiplying the difference signal by a convergence factor, updates an impulse response value by using the product signal, and outputs the updated impulse response value to the convolution arithmetic unit.

Abstract: A digital communication receiver includes a blind equalizer using the Constant Modulus Algorithm (CMA) to compensate for channel transmission distortion in digital communication systems. Improved CMA performance is obtained by using a partial trellis decoder to predict 1 bit or 2 bits of the corresponding 3-bit transmitted symbol. The predicted bits from the partial trellis decoder are used to reduce the effective number of symbols in the source alphabet, which reduces steady state jitter of the CMA algorithm. Specifically, the received input signal to the CMA error calculation is shifted up or down by a computed delta (&Dgr;), in accordance with the predicted bit(s). In addition, a different constant gamma (&ggr;), for the CMA error calculation is selected in accordance with the predicted bit(s).

Abstract: A method for canceling echo. A threshold value signal is set to a preset value and the threshold value signal and a medium value signal are keep at a level of the preset value. Diagnosis is made as to whether a difference between an echo signal (I) and the medium value signal (I-1) is more than a positive threshold value or less than a negative threshold value. When the difference between an echo signal (I) and the medium value signal (I-1) is more than the positive threshold value, the level of the medium value signal (I) is increased by an upraise speed and the level of the threshold value signal (I) is increased by a first speed faster than the upraise speed. When the difference between the echo signal (I) and the medium value signal (I-1) is less than the negative threshold value, the level of the medium value signal (I) is decreased by a drop speed and the level of the threshold value signal (I) is decreased by a second speed faster than the drop speed.

Abstract: A method of training a recursive filter comprises updating recursive parameters of the filter with delta values computed from sampling of an error signal and obtaining derivative terms of the output signal with respect to the parameters. Adaptation using derivatives for FIR filters is known, but this technique was not available for IIR filters because of the unavailability of the derivative values. However, when it is realized that the derivative itself is recursive, a parameter derivative function is obtained which will produce derivative terms for updating the filter parameters with acceptable filter performance.