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: According to an embodiment of present invention, an algorithm for computing static pre-equalizer coefficients, comprises the steps of determining a length of algorithm iterations; calculating a feedforward coefficient vector associated with a feedforward equalizer; calculating a pre-equalizer coefficient vector associated with a pre-equalizer filter; and performing the steps of calculating for the length of the algorithm iterations; wherein a mean square of an error between an output sequence and a transmitted digital input sequence is minimized.

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: Quick real-time identification and estimation of a time-non-varying or time-varying system. A new H? evaluation criterion is determined, a fast algorithm for a modified H? filter based on the criterion is developed, and a quick time-varying system identifying method according to the fast H? filtering algorithm is provided. By the fast H? filtering algorithm, a time-varying system sharply varying can be traced with an amount of calculation O(N) per unit time step. The algorithm completely agrees with a fast Kalman filtering algorithm at the extreme of the upper limit value. If the estimate of impulse response is determined, a pseudo-echo is sequentially determined from the estimate and subtracted from the real echo to cancel the echo. Thus an echo chancellor is realized.

Abstract: A demodulation method, and apparatus, for reproducing data from a received signal. The demodulation method includes the steps of: receiving a signal including a series of frames each containing a training signal for automatic equalization processing and a data signal; controlling an amplitude of the received signal by AGC processing so as to make the amplitude a predetermined level; conducting demodulation processing on the signal subjected to the AGC processing; periodically conducting automatic equalization processing so as to adapt the modulated signal to characteristics of the signal transmission path based on each training signal of the frame; and setting values of a time constant of the AGC processing and an update repetition period of the automatic equalization processing so as to satisfy a predetermined relation.

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: An apparatus processes an equalizer output signal. The equalizer output signal is formed by transmitting an alternate mark inversion input signal over a channel and passing the transmitted signal through an adaptive equalizer. The apparatus includes a correlator circuit block that detects an incorrect convergence of the adaptive equalizer and outputs a correlator output signal. A corrector filter receives the equalizer output signal and the correlator output signal, and applies a correction to the equalizer output signal based on the correlator output signal, to form a corrected signal that is substantially a time delayed copy of the input signal.

Abstract: A single-axis receiver processing, for example, complex vestigial sideband modulated signals with an equalizer with forward and feedback filters. Forward and feedback filters have parameters that are initialized and adapted to steady state operation. Adaptive equalization employs linear predictive filtering and error term generation based on various cost criteria. Adaptive equalization includes recursive update of parameters for forward and feedback filtering as operation changes between linear and decision-feedback equalization of either single or multi-channel signals. An adaptive, linear predictive filter generates real-valued parameters that are employed to set the parameters of the feedback filter. In an initialization mode, filter parameters are set via a linear prediction filter to approximate the inverse of the channel's impulse/frequency response and a constant modulus error term for adaptation of the filter parameters.

Abstract: A receiver for improving the performance of conventional Discrete Multitone Modulation (DMT) based Asymmetric Digital Subscriber Line (ADSL) modems, in the presence of noise and/or interference. A demodulator having an FFT followed by a single-tap-per-bin frequency-domain equalizer is augmented by an additional data-path utilizing windowing or pulse shaping. Windowing is done independently for each symbol over the orthogonality interval and efficiently in the time domain or frequency domain. A decision feedback equalizer at the output of the windowed data-path cancels inter-bin-interference created by windowing.

Abstract: A method equalizes a received signal in an optical communications system. The received signal is passed through an analog delay line where it is tapped to generate a set of delayed copies of the received signal. In a set of analog multipliers, each delayed copy of the received signal is multiplied by a corresponding weight to generate a set of weighted signals that are then summed to produce an output signal. The output signal is thresholded and subtracted from the output signal to produce an error signal, which is periodically sampled. In a digital weight updating circuit, the weights are produced from digitized versions of the sampled error signal and samples of the delayed copies of the received signal.

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 adaptive equalizer processes an input signal that includes noise, pre-cursor intersymbol interference, and post-cursor intersymbol interference. The adaptive equalizer includes a feedforward filter which reduces the pre-cursor intersymbol interference and whitens the noise, a feedback filter which detects the post-cursor intersymbol interference in a signal that corresponds to the input signal, and circuitry which removes the detected post-cursor intersymbol interference from the input signal. The feedforward filter includes separate first and second coefficients. The first coefficients reduce the pre-cursor intersymbol interference and the second coefficients whiten the noise.

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: An apparatus and method suitable for producing a set of filter coefficients are provided. Sequences of samples of first and second signals are received by the apparatus. Each received sample is processed to update a set of auto-correlation data elements and cross-correlation data elements. The set of auto-correlation data elements corresponds to a compressed version of a corresponding auto-correlation matrix data structure. A scheduling controller generates a scheduling signal including a succession of scheduling commands, the scheduling command indicating that a new set of filter coefficients is to be computed. A new scheduling command is issued when at least two samples of first and second signals are received subsequent to a previously issued scheduling command. In response to a scheduling command, the auto-correlation data elements are processed to generate the auto-correlation matrix data structure.

Abstract: A programmable driver/equalizer with an alterable FIR enables the equalization of serial links or other transmission systems to adapt to a variety of transmission media, specifically, intersymbol interference (ISI). Current mode differential drive circuits are coupled to a transmission media via a Finite Impulse Response (FIR) filter operating in the Z transform mode. The FIR filter includes A and B coefficient setting circuit, and is coupled to the drivers. The driver circuit also includes A coefficient level driver compensation and B coefficient level driver compensation to reduce self-induced ISI from the driver while the filter coefficients are activated. The coefficient setting circuit receive a combination of control bits to select the appropriate response for the driver to the various transmission media parameters. Adjustments to the driver output current are made at data run lengths exceeding certain values and subsequent adjustments are made for data run lengths exceeding larger values.

Abstract: A decision feed back equalizer provides two feedback routines for one symbol-preceding decided data. The signal space is separated into decidable areas and uncertainty areas. Within each of decidable area the distance between any point and the signal point is small. Within each uncertainty area the distance between any point and the signal point is large. If the symbol-preceding equalized signal is in a decidable area, the piece of decided data is fed back, as it is presumed to be correct. If the symbol-preceding equalized signal exists in an uncertainty area, the piece of decided data is not fed back, as it is presumed to be in error. Here the one piece data is selected from all the predicted symbol-preceding decided data so that the instant equalized data is now in the decidable area. This piece of decided data is now fed back.

Abstract: A filter for determining the cross-correlation of optically transmitted signals is disclosed which comprises a ladder structure of amplifier cells 3 and multipliers 4. Such a filter structure is integrated into a receiver for optical signals, the receiver being adaptable to the conditions of the transmission link through a recursive algorithm.

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: An adaptive filter includes: a plurality of delay circuits which receive a first signal, a second signal obtained by delaying the first signal, and a tap coefficient, and generates a first delay signal obtained by delaying the first signal, and a second delay signal obtained by delaying the second signal, and based upon the first delay signal and the tap coefficient, calculates a set point signal for each of the delay circuits; an addition circuit which adds the set point signals for a set point signal as a whole; and a coefficient calculation circuit which is placed in each of the delay circuits and which sequentially selects the second delay signals, and based upon the selected second delay signal and the set point signal as a whole, sequentially calculates the tap coefficients for the respective delay circuits.

Abstract: A high-speed broadband, wireline modem including an adaptive equalizer having both a training mode and a decision-directed non-training mode. The adaptive equalizer comprising at least one of a forward path coupled to receive signal samples, the forward path including a forward filter and a decision element, and a feedback path coupled between an output of the decision element and an input of the decision element, the feedback path including a feedback filter; and means for adapting the one of said forward filter and said feedback filter based on a least squares error criterion performed substantially according to a predetermined algorithm.

Abstract: A training circuit for training a tap coefficient of an adaptive equalizer that performs error calculations in a training process and updating of the tap coefficient in a time area and thereby makes the tap coefficient converge stably in a short time.

Abstract: A real/complex dual combination channel equalizer is provided, which is capable of minimizing the complexity of a channel equalizer while maintaining the performance of a real and complex mode. The real/complex dual combination channel equalizer includes a combination of elements for inputting real data and imaginary data to a channel select unit, outputting data selected by the channel select unit, yielding real filter output data and imaginary filter output data by multiplying tap coefficients to the outputted data and performing operations, selecting data among the yielded data, outputting a final data of the real filter by subtracting the value outputted from a real decision orientation filter unit from the selected data, outputting real/complex decision data and an error by the outputted data, and renewing the tap coefficient by the data outputted from the channel select unit and the error.

Abstract: A method of adaptive multi-modulus equalization for an equalizer. A cost from a cost function is calculated according to a constant modulus algorithm (CMA). Equalizer coefficients are updated according to the cost, and modulus of each region is determined for a multi-modulus algorithm (MMA). The equalizer is switched to use the MMA when the cost reaches a first threshold. The MMA comprises several stages determined by thresholds, and the number of regions increases in every stage. The cost is calculated according to the MMA and modulus of each region. The equalizer coefficients are updated according to the cost, and modulus of each region is determined for the subsequent stage of the MMA. The equalizer is switched to the subsequent stage of the MMA when the MMA cost function output reaches the threshold corresponding to the current stage. The steps are repeated until the cost reaches a preset value. Then, the number of regions and equalizer coefficients are fixed to equalize the input signal.

Abstract: The detection of symbols in digital burst transmissions is improved by an equalizer (30) that initializes the values of only a subset of its filter coefficients (C, D), and thereafter begins to decode the symbols using an iterative, decision directed algorithm that determines all of its filter coefficients. This can advantageously provide an acceptable trade-off between computational complexity and equalizer performance.

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.

Abstract: The invention relates to an apparatus and method of detecting a sequence of information symbols from a first signal subjected to inter-symbol interference. The method is performed as one or more signal processing paths are each being adapted to setting each symbol in the sequence to a value. When an uncertain decision has been taken in a given signal processing path, the signal processing path is divided into two. After setting a number of symbols, the sequence of information symbols is selected. The threshold used to determine whether a symbol is certain or not is adjusted in accordance with an estimate of the noise. The number of symbol errors when detecting a signal subjected to inter-symbol interference, e.g. in a mobile station, is reduced, and therefore the performance of the receiver is improved. Since unnecessary calculations are minimized, computational complexity is reduced which, in turn, reduces the power consumption.

Abstract: A demodulation circuit for demodulating a multilevel digital modulation signal. Only an in-phase or quadrature component of the training signal generated in the demodulation circuit is used for equalization convergence. Tap coefficients of an equalizing training equalizer are updated by supplying either the in-phase or quadrature component so that the calculation amount can be reduced. When the equalization convergence is established, the obtained tap coefficients are rotated in phase and set to a data reproduction equalizer.

Abstract: A feedforward equalizer for DFE based detector is provided comprising a digital to analog converter to convert an analog signal to a digital signal. A feedforward equalizer comprises a high-pass filter and is responsive to the input circuit. The high-pass filter has a low cutoff frequency, has a relatively flat response and has high attenuation at low frequencies. A decision feedback equalizer comprises a decision circuit responsive to the feedforward equalizer, and a feedback filter is responsive to the decision circuit. The decision circuit is also responsive to the feedback filter.

Abstract: A modem receiver has an equalizer which receives an input signal, an adaptive compensator, and a decision block which generates a decision related to the input signal. A difference between an output of the equalizer and a compensating signal output of the adaptive compensator provides an input for the decision block. A difference between the output of the equalizer and an output of the decision block provides an error estimation signal which is a first input to the adaptive compensator. A difference between the input to the decision block and the output of the decision block provides an error adaptation signal which is a second input to the adaptive compensator.

Abstract: This invention provides a method for initializing the filter coefficients of a hybrid frequency-time domain adaptive equalizer device implementing frequency domain (FD) filter equalization in a forward path and a time domain (TD) filter equalization in a feedback path, with each filter unit having a plurality of adaptable filter taps. Preferably, in initializing the equalizer, the relation 1 + F G = H is obeyed “where (G) is the forward FD equalizer taps, (F) is” the Fast Fourier Transform of the feedback TD equalizer taps, and (H) is the estimated (frequency domain) response of the communications channel.

Abstract: For processing samples of a signal received via a channel represented by an impulse response of W?1 coefficients, the method comprises: determining the W roots of the Z-transform of the channel impulse response; producing an intermediate signal by equalizing the received signal by a zero-forcing method or the like based on an impulse response whose Z-transform is a Z?1 polynomial of degree W-p having as roots those of the W roots which are furthest from the unit circle of the complex plane; and then obtaining estimations of the transmitted signal symbols by applying a Viterbi-type equalization method or the like based on an impulse response whose Z-transform is a Z?1 polynomial of degree p having as roots those of the W roots which are nearest to the unit circle.

Abstract: A digital signal processing apparatus for reading out a signal from a recording medium of a digital recording apparatus of high-frequency cutoff type is provided which includes: a read head 2 for reading out the recorded signal from the recording medium 2; a low-pass filter 3 for removing high frequency noise from the output signal of the read head 2; an analog/digital converter 4 for converting the analog reproduced signal filtered by the low-pass filter 3 to a digital reproduced signal; an FIR filter 5 for filtering the digital reproduced signal with the use of an adaptive equalization coefficient so that the frequency response of the signal is equalized with the partial response which has a frequency characteristic suited for reading the reproduced signal; an adaptive equalization coefficient setting device 6 and a partial response temporal judgment device 7 for determining the adaptive equalization coefficient; and a Viterbi decoder 8 for decoding a partial response equalized data released from the FIR fi

Abstract: A signal processing circuit for a digital signal receiving system includes an equalization unit for performing waveform-equalization of a received signal, the equalization unit including a backward equalizer constituted by a feedback loop; and a phase compensation unit for performing phase compensation of the received signal. The phase compensation unit is arranged upstream of the backward equalizer. Thus, it is possible to avoid the deterioration of the phase compensation ability of the phase compensation unit by the affection from the echo in the transmission path.

Abstract: A baseband signal processing method for calculating minimum phase prefilters. The method preferably utilizes a spectral factorization approach. The prefilter is constructed from a minimum phase version of an estimated channel impulse response. Complex valued filter coefficients are determined by solving a system of linear equations for each iteration of the spectral factorization. The constructed prefilters provide equalization in both the forward and backward directions.

Abstract: A decision feedback equalizer and a method for updating tap coefficients of the decision feedback equalizer are provided. The decision feedback equalizer (DFE) includes a first equalizer for reducing channel interference noise in an input signal, a second equalizer for generating a feedback signal for reducing remaining interference noise, and an update controller for controlling the first and the second equalizers so as to freeze the second equalizer when the first equalizer is updated and to freeze the first equalizer when the second equalizer is updated. The tap coefficients of the feedforward equalizer and the feedback equalizer are alternately initialized. It then becomes possible to stably converge the levels of respective tap coefficients without diverging them and to reduce the power used for updating the tap coefficients in a steady state.

Abstract: A semiconductor circuit includes a decision feedback equalizer (DFE) for waveform-equalizing an input signal and generating a waveform-equalized input signal. The DFE compares the waveform-equalized signal with a predetermined reference voltage to generate a decision signal having first and second decision values and an error signal which lies between the waveform-equalized signal and the decision signal. A dispersion value calculator is connected to the DFE, calculates first and second dispersion values of the first and second decision values of the decision signal using the error signal, and produces a compensation signal using the first and second dispersion values. An asymmetry compensator is connected to the DFE and the dispersion value calculator. The asymmetry compensator receives the input signal and corrects an asymmetry in the input signal in accordance with the compensation signal and supplies the corrected input signal to the DFE.

Abstract: A method and apparatus for decision-directed adaptation for coded modulation is presented in which a modulation decoder and data re-encoder are used to create an estimated version of a received signal based on the use of a reliable modulation code. An adaptive equalizer is used to process the received signal, and the differences between the received signal passed through the adaptive equalizer and the estimated version are used by the adaptive equalizer to set the equalizer coefficients. A variety of adaptation algorithms can be used including Least Mean Squares (LMS) and Recursive Least Squares (RLS).

Abstract: In an adaptive audio equalizer apparatus, a signal that is output from an adaptive filter 13 is fed to a speaker and to a delaying unit. The signal fed to the delaying unit is delayed for a predetermined time, and is multiplied by a scaling factor in a multiplying unit. A first calculation unit calculates a difference between an output of a microphone and a target response signal, and outputs the result as an error signal. A second calculation unit adds the output of the multiplying unit to the error signal, and outputs the result to a filter coefficient setting unit of the adaptive filter.

Abstract: A computationally efficient method for computing the filter coefficients for a prefilter in a decision feedback equalizer solves linear equations using a fast Toeplitz algorithm. Computations performed to compute the filter coefficients for the right half burst may be used to compute the prefilter for the left hand burst, thereby reducing the number of computations. Also, a square root-free algorithm may be used to solve the system of linear equations, further reducing computational complexity.

Abstract: An equalizing apparatus includes an equalizer which has a plurality of adjustable tap weights that equalizes a received signal based on values of the adjustable tap weights, a tap weight update calculation unit coupled to the equalizer and which determines tap weight updates for use in adjusting the tap weights during operation of the equalizer, an offset memory that stores one or more tap weight update offset values and a summer coupled to the tap weight update calculation unit and to the offset memory. The summer combines each of the tap weight updates with one of the tap weight update offset values to produce modified tap weight updates which, in turn, are provided to the equalizer to adjust the tap weights.

Abstract: The invention relates to a method and circuit arrangement for compensating for an interference component included in a signal in a digital transmission system such as a mobile communication system, radio system or television system. In the method according to the invention a decision error representing an interfering signal is classified using an adaptive method, and a table is drawn up of interference estimates corresponding to classified decision errors. So it is obtained on the basis of previous decision errors an interference estimate corresponding to the next signal sample and that estimate is subtracted from the signal sample before the current decision. The adaptive method is preferably a neural method such as a self-organizing map. With the method according to the invention it is possible to compensate for an interference component in a signal regardless of the interference source assuming that the interfering signal has internal correlation.

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 invention is a method and apparatus for determining the coefficients to be used by a time domain equalization circuit in a receiver in a communication system using discrete multitone (DMT) technology. The method and apparatus is particularly efficient in that the order of the channel response equation after channel shortening is less than or equal to the allotted cyclic prefix length utilized in the applicable communication protocol being used. Further, the invention includes an efficient method and apparatus for determining symbol synchronization for the time domain equalization.

Abstract: A system, device, and method for time-domain equalizer (TEQ) training determines the TEQ order and TEQ coefficients by applying the multichannel Levinson algorithm for auto-regressive moving average (ARMA) modeling of the channel impulse response. Specifically, the TEQ is trained based upon a received training signal. The received training signal and knowledge of the transmitted training signal are used to derive an autocorrelation matrix that is used in formulating the multichannel ARMA model. The parameters of the multichannel ARMA model are estimated via a recursive procedure using the multichannel Levinson algorithm. Starting from a sufficiently high-order model with a fixed pole-zero difference, the TEQ coefficients corresponding to a low-order model are derived from those of a high-order model.

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 waveform equalizer which can improve a quality of a read signal by certainly correcting asymmetry in the read signal and a data reproducing apparatus using the waveform equalizer. The waveform equalizer performs a waveform equalization to the read signal and is constructed by: inclination discriminating means for detecting an ascending slope portion and a descending slope portion of the read signal; a filter for performing a filtering process to the read signal; and characteristics switching means for switching frequency characteristics of the filter in accordance with a discrimination output of the inclination discriminating means.

Abstract: A delayed decision feedback Log-MAP equalizer generates optimal soft information at the bit level for MPSK and MDPSK modulations over multipath fading channels. The algorithms have a forward-backward recursion structure similar to conventional Log-MAP algorithms. To reduce complexity, the forward recursion incorporates the techniques of delayed decision feedback sequence estimation to construct a reduced state space. The backward recursion then takes this state space as given, in that it does not hypothesize feedbacks, and computes required soft information.

Abstract: A signal processor includes an adaptive filter that generates an output signal as a function of an input signal and of weighting coefficients stored in a memory. An error detector generates an error signal as a function of the output signal. The error signal is used in adjusting the weighting coefficients. A data processor adjusts the number of weighting coefficients of the adaptive filter and system resources used for storing the weighting coefficients as a function of a characteristic of the weighting coefficients.

Abstract: A method for an adaptive equalization apparatus in a multiple-link hopping radio system includes hopping among a plurality of radio links to receive variable-length bursts of radio signals on the plurality of radio links and equalizing amplitude and phase variations of a slow channel for each radio link from a received burst on the radio link. Further, the method includes storing the estimated tap coefficients pertinent to each radio link and using the tap weights of the current burst of the radio link to reliably pre-compensate the channel amplitude and phase distortion of a next received burst on the radio link.

Abstract: A technique for training an equalizer of an ADSL Annex C transceiver is provided. The technique defines the points in the transmission sequence at which equalizer training should take place, and what symbols comprising the transmission should be used for training. The technique avoids the effect of strong TCM-ISDN cross-talk noise. The equalizer is trained so as to achieve optimal communication channel performance. During the TEQ training phase, the cyclic prefix is not introduced. The TEQ is trained only on FEXT symbols except the FEXT symbols adjacent to NEXT symbols. In the early phase of FEQ training, the cyclic prefix is not introduced. The FEQ is trained on FEXT symbols only except the FEXT symbols adjacent to NEXT symbols. During a late training phase after the cyclic prefix is introduced, the training of the FEQ is continued by using signals received on all FEXT symbols only.