Abstract: An adaptive DMT modem determines, as part of a startup sequence, subscriber loop conditions. One or more loop condition adaptation operations are then performed to reduce the complexity of the filters employed in the modem, to reduce the power dissipated in the modem, to optimize performance of the modem or to reduce the likelihood of EMI being radiated into the premises of a subscriber.

Abstract: A timing phase acquisition method and device for burst modems includes an receiver designed to initialize an equalizer filter by matching clock of the equalizer filter of the receiver with the phase of the received signal.

Abstract: An object is to obtain an adaptive equalizer device and an adaptive equalizer controlling method which provide an ideal output signal certainly and in a shorter time. The adaptive equalizer device includes an adaptive equalizer (4a) receiving a signal indicating a symbol, for filtering the signal indicating the symbol to adaptively remove intersymbol interference by selectively using one of a plurality of preset algorithms, and a rotation detecting circuit (19) receiving an output signal (y(n)) from the adaptive equalizer, for detecting rotation of the symbol of the output signal (y(n)), wherein the adaptive equalizer (4a) selects one of the plurality of algorithms in accordance with detected result flags (&agr;, &bgr;) from the rotation detecting circuit (19).

Abstract: Even with an asymmetric reproduction waveform, or under the noisy environment, an accurate equalizing error is found and temporary judgment value for adaptively controlling the coefficient of FIR filter is accurately outputted. A waveform equalizer for equalizing a waveform of a reproduction signal obtained by reproducing marks and non-marks recorded on a recording medium is provided. The equalizer includes: a detector which detects asymmetry of the reproduction signal arising from physical profiles of the marks and non-marks and outputs a detection signal representing asymmetric amount; a discriminator which discriminates the marks and non-marks based on the reproduction signal to output a discriminated signal; and a selector which calculates a first and second coefficient multiplied by the reproduction signal of the marks and non-marks based on the detection signal, which are different. The selector selects one of the first and second coefficients based on the discriminating signal.

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: The present invention can be used to perform training using secondary data. In one embodiment the present invention includes receiving a communications signal of a communications channel, where the communications signal has a primary and a secondary data segment both communicating information, and determining a parameter of the communications channel using the secondary data segment. In another embodiment of the present invention, the communications signal can also include a training segment containing a known training signal.

Abstract: A method and apparatus is disclosed to overcome the effects of intersymbol interference during data transmission. Overcoming the effects of intersymbol interference makes possible higher data transmission rates for a given error rate. In one embodiment a receiver-transmitter pair is configured with a precode filter at the transmit side and a feed forward filter and a feedback filter at the receive side. Filter coefficients are calculated to reduce the undesirable effects of the channel, such as intersymbol interference. In one embodiment a training process occurs with the feedforward filter and a feedback filter, such that the first N coefficients of the feedback filter are set to zero. Thereafter, the coefficients of the feedforward filter are subject to spectral factorization and separated into minimum phase roots and maximum phase roots. The minimum phase roots comprise the precode filter coefficients and the maximum phase roots are established as feedforward filter coefficients.

Abstract: A method for estimating noise auto-correlation for an equalizer includes the step of estimating a noise auto-correlation and weighting the estimated noise auto-correlation by a selected weighted window.

Abstract: A digital subscriber line receiving apparatus having resilience to HAM radio interference noise and a method therefore includes an automatic gain controller for amplifying a received signal level to a signal level having a predetermined magnitude, a timing recoverer for recovering timing from a signal output from the automatic gain controller, an equalizer for removing interference noise from a signal received from the timing recoverer, a slicer for detecting a digital value from a signal received from the equalizer, an initialization controller for resetting the receiving apparatus and initializing the automatic gain controller, the timing recoverer, and the equalizer when the signal to noise ratio of the output signal of the equalizer is smaller than a reference signal to noise ratio, and an update controller for preventing automatic gain controller and equalizer from being initialized by the initialization controller when HAM radio interference noise is detected in the received signal.

Abstract: An apparatus and method for minimizing nonlinear distortions in computer system communications where the upstream signal from a client modem to a server modem is periodically sampled, the samples being utilized by a non-linear decision feedback equalizer to periodically produce sets of equalizer coefficients. The equalizer coefficients are sent via an independent communication channel downstream to the client modem to continuously update distortion-minimizing adjustments to the client modem's upstream output. The samples are taken periodically in order to update the non-linear decision feedback equalizer so that it can provide appropriate equalizer coefficients for the changing characteristics of the upstream signal. The method includes identifying, by a first communication system, nonlinear equalization parameters to be used by a second communication system to minimize nonlinear distortions on a primary communication channel.

Abstract: A communication circuitry for providing effective detection of tones in a communication network. The communication circuitry comprises filtration circuitry, summation circuitry, detection circuitry, and interface circuitry. The filtration circuitry is coupled to the summation circuitry and is configured to receive a communication, process the communication to estimate an echo return in the communication, and provide the estimated echo return to the summation circuitry. The summation circuitry is coupled to the detection circuitry and configured to receive the estimated echo return from the filtration circuitry, receive a tone transmission and echo return from the interface circuitry, subtract the estimated echo return from the tone transmission to generate a substantially pure tone, and provide the substantially pure tone to the detection circuitry. The detection circuitry is configured to receive the substantially pure tone and process the substantially pure tone to perform a programmed operation.

Abstract: K sub-carrier components from an FFT circuit 101 are output to a sub-carrier demodulating circuit 102. The sub-carrier demodulating circuit 102 demodulates the k sub-carrier components and outputs a transmitted information bit to an output terminal 104. Further, the sub-carrier demodulating circuit 102 outputs distortion correction coefficients C(1)-C(k) that are used during demodulation of each sub-carrier, each of sub-carrier components S(1)-S(k) in which distortion is corrected, and determination signal points R(1)-R(k) for each sub-carrier to a phase estimating circuit 103. The phase estimating circuit 103 receives C(1)-C(k), R(1)-R(k) and S(1)-S(k), and outputs a phase error signal to an output terminal 105. The phase error signal output to the output terminal 105 is integrated by a loop filter and so forth, and is fed back to a local oscillator, and thereby, phase tracking is conducted.

Abstract: The present invention proposes a signal processing device for processing received signals having been transmitted via a transmission channel, wherein said received signals (y) are supplied to an estimation means (2) adapted to estimated a channel impulse response function (h) of said transmission channel (1), with said estimation means being adapted to perform a Linear-Minimum-Mean-Square-Error-(LMMSE-) processing based on a training sequence (TR_SEQ) transmitted together with received samples contained in said received signals (y), and a noise and interference energy estimate represented by a noise and interference variance (&dgr;2) as well as an impulse response energy distribution estimate represented by an estimated parameter covariance matrix (Chh) for said channel (1). The present invention also proposes a corresponding method for signal processing.

Abstract: Channel estimation is performed for a received signal, wherein different parts of the received signals are modulated using different modulation types. The channel is first estimated based on part of a received signal modulated with a first modulation type. This channel estimate is then transformed into a channel estimate corresponding to at least a second modulation type.

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.

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 low-complexity frequency domain equalizer having fast re-lock capabilities for use in a DMT transceiver. The transceiver includes a modulator that utilizes DMT modulation methods for communication over the transmission medium, a demodulator for receiving DMT signals, and a channel equalizer that operates in the frequency domain. The equalizer models the channel characteristic as an FIR filter that distorts the signal transmitted over the channel. The channel distortion is removed at the receiver by performing a de-convolution of the channel response via a frequency domain multiplication. After a period of inactivity of the receiver, the equalizer taps are updated by the use of a fast re-lock method so that the previously trained equalizer may be utilized. An undetermined sampling time offset is determined and is used to update the equalizer coefficients such that a full retraining of the equalizer is not required.

Abstract: A system and method for enhancing the performance of an equalizer in a modem. Multiple techniques are disclosed which improve the modem performance. A first technique uses stored parameters for each burst from each remote site to demodulate a received data stream. A second technique compensates for the gain droop caused by storing parameters across each burst. A third technique minimizes errors caused by adapting the equalizer coefficients for each data burst by analyzing the SN ratio and error rate of the received burst. A fourth technique improves the convergence of the equalizer by using a two-part preamble, whereby both parts are transmitted using different modulation techniques. A fifth technique is provided which performs a soft reset of the modem without performing a complete reset of the modem. A sixth technique determines a modem adaptation factor based on the expected modulation type of an incoming burst transmission.

Abstract: The present invention is directed to a reconfigurable finite impulse response (FIR) filter that processes data tap values with canonical sign digit (CSD) coefficients. The FIR filter according to the present invention includes a shift group associated with at least one of the data tap values and at least one of the CSD coefficients. Each shift group includes a plurality of parallel paths. Each parallel path shifts the data tap value according to a bit pair of the coefficient to produce a weighted product. An adder is also included that combines the weighted products from each of the shift groups.

Abstract: A method for training a time-domain equalizer having at least one coefficient that includes estimating a channel, initializing the at least one coefficient of the time-domain equalizer, updating the at least one coefficient of the time-domain equalizer with the estimated channel, retaining the updated estimated channel, fixing the updated value of the at least one coefficient of the time-domain equalizer for at least a one-symbol duration, calculating a modulated symbol based on an output of the time-domain equalizer, calculating a second value for the estimated channel based on the modulated symbol, setting the estimated channel to the second value, and repeating the step of updating the time-domain equalizer through the step of setting the estimated channel to the second value until a predetermined condition has been met.

Abstract: In the Global System for Mobile Communication (GSM), an improvement in estimating the channel impulse response (CIR) is obtained by starting up two channel estimators using the midamble and obtaining channel estimations with split burst. The two estimations are then cross-correlated to obtain the estimated CIR.

Abstract: The present invention provides methods and systems for providing efficient use of available CPU cycles that may be achieved by reducing the length of the echo canceller filter. An echo canceller's adaptive algorithm may be implemented to reach a lower mean squared error due to the increased training time allowed by cycle reduction. Total power consumption of a processor may be reduced by reducing the number of multiplications (and/or other operations) that may be performed by an echo canceller filter for each time iteration. This may be achieved by not performing multiplications on taps that are not being used. For higher rates, a shorter echo canceller filter may be implemented by transmitting less energy in low frequencies. As a result, the temporal length of the echo channel may be shortened thereby reducing the total number of taps needed to model the channel. The present invention may further enhance performance by reducing CPU cycles and allowing for more echo canceller training time.

Abstract: A simplified method for sampling timing adjustment and frequency offset estimation in a TDMA cellular PCS environment using &pgr;/4 - shifted DQPSK comprises the steps of oversampling a received signal resulting from transmission of sequences of complex-valued symbols at a rate N times the symbol rate thereof so as to produce N sets of samples, comparing for each set of samples the differential phase angle between successively received complex-valued symbols, and determining which set of the N sets of samples has differential phase angles closest to ideal values to thereby obtain an optimal sampling timing. The differential phase angles are measured by multiplying a complex conjugate of a received complex-valued symbol and a succeeding symbol to produce a comparison vector having an angle equal to the differential phase angle between the received complex-valued symbol and the succeeding symbol. The differential phase angles are optionally rotated so that the angle thereof is between 0° and 90°.

Abstract: The invention relates to a method and apparatus for implementing the training phase of adaptive channel equalization on a digital communications path passing signals modulated using QAM or CAP in a system configuration where the transmitter or the receiver or both of them include bandstop filters that form at least one stop band on the signal spectrum. In the method, the training of the adaptive equalizers is performed utilizing a sparse symbol constellation, wherein the number of symbol states (S) is reduced from that of the symbol constellation used in the normal data transmission state. The benefit of the sparse symbol constellation is that the strong intersymbol interference caused by a stop band falling on the signal spectrum can thus be reduced to such a low level in regard to the signal power that a linear equalizer may be used to reach a situation, wherein the proportion of correct symbol recovery decisions becomes sufficiently high for the training of a decision-feedback equalizer.

Abstract: Channel estimation is performed in a mobile communications system. In order to reduce the computational load, and thereby decrease power consumption, a polynomial expression is used to approximate a Minimum Mean Squared Error (MMSE) based channel estimator with respect to Doppler and SIR variations. A lookup table may be used to supply coefficients for the polynomial expression. In another aspect of the invention, coefficients for the polynomial expression may, themselves, be approximated by a second polynomial expression.

Abstract: A method of estimating channel impulse response in a signal transmitted over a channel in a communication system is described. The channel estimator obtains a priori knowledge about the transmitted signal, and then uses the transmitted signal and the a priori knowledge to choose an estimate of channel impulse response which minimizes the expected distance between the transmitted signal and a reconstructed signal. The expected distance that is minimized is a cost function, represented by E{∥r−Bh∥2r}. By using soft decision feedback, the invention minimizes erroneous decision feedback which can cause error propagation. The decisions are usually in the form of log likelihood ratios (LLR).

Abstract: A method for fast training of equalizers in a DMT system begins by normalizing the incoming receive signal (&ngr;) via steps (108-116). By normalizing the signal, the convergence rate of the training algorithm becomes relatively independent of channel line length so that long line lengths may converge to optimal equalizer coefficients in short time periods. The method also iteratively adjusts the filter coefficients w over time by using an adaptive gain vector &mgr; that is updated on a component-by-component basis on each iteration via steps (114-146). By allowing each component of the vector &mgr; to iteratively adapt independent of all other components in the vector &mgr; based upon the binary sign bit of both real and imaginary components of frequency domain gradient vectors G, a convergence to optimal equalizer filter coefficients will occur in a short period of time.

Abstract: A method and an apparatus for filtering an audio signal include the steps of making the audio signal available in digitalized form, wherein the duration of the sampling interval is half or less than half of the period duration of the highest frequency to be expected in the audio signal, a digitalized impulse response is made available in accordance with the desired filtering effect, and a convolution sum is formed from the samples of the impulse response and the samples of the audio signal, wherein, (i) several adjacent samples of the impulse response define a corresponding interval within at least a time portion of the impulse response which is shorter than the impulse response, (ii) within the interval defined in this manner, the samples of the impulse response corresponding to the interval defined in this manner are equated to a value which is a function of one or more of the samples of the digitalized impulse response falling within the interval defined in this manner, and (iii) the steps (i) and (ii) are

Abstract: The present invention employs training sequence hopping in a cellular communications system to permit tight frequency reuse by minimizing or avoiding co-channel interference and/or other signal degrading phenomena. During a first portion of a transmission, e.g., a first burst, a first training sequence is used. During a second portion of that transmission, e.g., a second burst, a second different training sequence is used. Different training sequences are used for different portions of the transmission according to a predetermined training sequence hopping pattern and with a predetermined number of training sequences. The portion during can extend for the length of a burst, for more than one burst, for a speech frame, or for some other time period.

Abstract: A client-side modem (20) having capability of improved data rate upstream communications via a codec (2) in a central office (CO) to an Internet service provider (ISP) (10) is disclosed. The disclosed modem (20) includes a digital signal processor (DSP) (14) that, for upstream communications, performs a bit mapping function (22) upon the digital data so that the digital words are mapped to slicing levels of analog-to-digital conversion circuitry (A/D) (6) in the central office codec (2); in addition, bit mapping function (22) preferably maps the digital signal into symbols having fewer bits each than the maximum defined by the number of slicing levels of the A/D (6). A pre-equalize filter function (26) applies an approximate inverse channel response filter (as measured during the training process) to the bit mapped digital data, to maximize the transmitted data rage over the dispersive analog subscriber loop (ASL), with consideration for an average transmit power maximum specification.

Abstract: Since a conventional adaptive equalizer handles an update quantity control parameter &mgr; as a fixed value, the update quantity control parameter &mgr; is not optimized adaptively to variation of a channel state and the BER characteristic is deteriorated.

Abstract: Method and apparatus that compensates for overestimation of channel SINR due to overfitting of an adaptive equalizer to the pilot portion of a received frame. The SINR of a wireless channel is estimated by adapting an adaptive equalizer using the pilot portion of the frame, applying the adaptive equalizer to a non-pilot portion of the frame, determining a parameter using the adaptive equalizer output, and estimating the SINR of the wireless channel using the parameter. The parameter can include, for example, the mean square error (MSE) or the bias of the equalizer output. The accuracy of the SINR estimate according to this aspect of the present invention is thereby improved (as compared to a SINR estimate based on parameters calculated during the pilot interval) because the adaptive equalizer is not overfit to the control or data portions of the frame.

Abstract: A frequency estimator is provided for use in a receiver of packetized data. The frequency estimator determines the signal frequency error ({circumflex over (ƒ)}o) with which a data packet has been received. It does this by relating data (gn) representative of some symbols to decisions (Sn) on those symbols. These symbols include not only training symbols (Nt) but also some information symbols (Nadd).

Abstract: Rapid identification and modeling of transmission media channel characteristics of a communications system using a correlation based technique. The technique provides a known training sequence used to generate a known quantity that operates on an observed or measured received signal, which is a function of the training sequence and the channel's impulse response, to give an estimate of the model of the channel. The technique decouples the training sequence from the observed or measured output, leaving the estimated impulse response. The impulse response of the transmission media channel is rapidly computed and processed to set the initial values of filter coefficients, for example, an echo canceller and an equalizer, in the communications system. Once the coefficients are initialized, if needed, a standard technique, such as least mean square (LMS) correlation, can be used to fine-tune the coefficients to converge on or model the transmission media channel's characteristics.

Abstract: A method and apparatus for reducing convergence time in a digital filter. When the digital filter is initially run, the coefficients in the digital filter are adjusted to reduce error in the output of the digital filter. When the adjusted coefficients meet a selected error level, these coefficients are stored in a memory and the digital filter filters data. The next time the digital filter is run, the stored coefficients are loaded into the digital filter and a number of iterations are run in which the coefficients are adjusted. Then, a determination is made as to whether the error level meets a threshold that may be the same as the selected error level. If the coefficients meet the threshold, the coefficients are stored in the memory and the filter is then used to filter data.

Abstract: A method and apparatus for estimating channel impulse response and data in a signal transmitted over a channel in a communication system. The channel impulse response is estimated uses correlative channel sounding, and then, using the estimated channel impulse response, the data in the signal is estimated. The output is then fed back to the channel impulse response estimator and the channel impulse estimation is repeated. The data estimation and channel impulse response estimation may be iterated a number of times.

Abstract: A method and apparatus are disclosed for improving channel equalization and level learning in a data communication system. The disclosed equalizer training process separately updates the feed forward filter (FFF) and the level adapter, to gain additional improvements in the training of the feed forward filter (FFF). The multi-step equalizer training process initially trains the feed forward filter (FFF) using a two-level signal EQTR(n) having an ideal value (Step One) to help converge the feed forward filter (FFF) to a certain level. Once the feed forward filter (FFF) reaches a certain level of convergence, the training circuitry is reconfigured during step two of the equalizer training process, to evaluate and update the actual level of the signal EQTR(n), to compensate for the channel. The determined weighting factors are applied to a low pass filter and the actual level of the signal EQTR(n), B(n), is calculated.

Abstract: A discrete multitone (DMT) receiver has a time domain equalizer (TDEQ) which is trained during initialization using an estimated channel impulse response (CIR). To eliminate wrap-around of the estimated CIR relative to the DMT frame alignment, and to remove leading zeroes, the estimated CIR is circularly advanced relative to the frame alignment for training the TDEQ. In subsequent communications of DMT symbols via the channel and the TDEQ, the equalized samples of the received DMT symbols are buffered to compensate, in a frame alignment of the received DMT symbols, for the advance of the estimated CIR in the initialization process and for a delay in the TDEQ.

Abstract: A radio signal that has been received from a channel is equalized by estimating the channel, generating an initial estimate of a DC-offset that has been introduced into the radio signal, and estimating a variance of the initial estimate of the DC-offset. The channel estimate, the initial estimate of the DC-offset and the estimated variance are then used to determine a most likely symbol represented by the radio signal. The initial estimate of the DC-offset that has been introduced into the radio signal may be determined from a training sequence portion of the received signal and a reference training sequence signal. For example, a least squares technique may be used to generate the DC-offset as well as the channel estimate and the variance of the initial estimate of the DC-offset. In an equalizer section, the DC-offset and variance values are taken into consideration when the most likely symbol represented by the radio signal is decided.

Abstract: Equalizer training is validated by techniques to detect false training. The validation techniques involve checking the unique time domain energy shaping of a plurality of taps of a trained equalizer of a modem for good training. The energy measure for each equalizer tap of the plurality of taps is approximated by summing the magnitude of a real component and a magnitude of an imaginary component of an equalizer coefficient corresponding to the equalizer tap. A tap with a highest approximate energy (i.e., the main tap) is then determined. Next, the main tap energy is compared to a main tap threshold and a predetermined guarding threshold. The plurality of taps are arranged in a time index ordered sequence. In determining the main tap threshold, a first energy sum of a beginning set of taps in the sequence is compared with a second energy sum of a last set of taps in the sequence. The lesser tap energy sum between the first energy sum and the second energy sum is the main tap threshold.

Abstract: A digital subscriber line communication system does not require the use of a plain old telephone service (POTS) splitter in the resident's home. Digital signal processing techniques are utilized to adapt to varying subscriber line conditions due to POTS telephone equipment. The digital signal processing techniques eliminate the need for a splitter by reducing susceptibility to distortion due to varying subscriber line characteristics. The digital subscriber line modem characterizes the subscriber line under a variety of conditions when the modem is in a non-idle mode. The digital subscriber line modem includes a control circuit which performs rapid retrain operation utilizing line characterization information.

Abstract: A method and system for performing a timing recovery acquisition of a sinusoidal preamble with reduced loop latency by effectively bypassing the FIR filter out of the timing loop. The method is implemented by estimating the FIR filter's (201) phase shift through knowing its coefficients a priori while also determining the phase error at the FIR filter's input. Because of the non-symmetric settings of the coefficients the phase shift through the FIR may not be zero. It reduces the latency of a non-data tracking timing loop by eliminating latency due to the FIR filter (201) itself. The coefficients can be either programmed in memory (205) or adapted as part of an LMS application. The estimate avoids the use of multiply operations, using add operations and a single divider. The estimated partial phase errors are then summed in a “FIR bypass-mode” phase detector (204) yielding total phase shift.

Abstract: A method and a system for configuring and initializing an equalizer. The equalizer equalizes a signal received over a propagation channel. The received signal includes symbols, a synchronizing signal and is formed by one or more rays including a dominant ray. The method comprising the following: (a) generating a multi-path intensity profile of the propagation channel from the received signal; (b) determining an arrival time of the dominant ray and a propagation channel response corresponding to the arrival time based on the multi-path intensity profile; (c) decimating the received signal synchronously with the arrival time of the dominant ray to provide signal samples to the equalizer; (d) configuring the equalizer based on the multi-path intensity profile to deactivate some of the coefficients of the equalizer; and (e) initializing active coefficients of the equalizer based on the propagation channel response.

Abstract: Analysis filter splits an input signal into M frequency bands. Sampling rate converter 102 downsamples split signals at a sampling rate of 1/M times. Training processing section 105 receives a known training signal and provides the training processing to the signal for the distortion correction. Sampling rate converter 103 upsamples a corrected signal. Synthesis filter 104 reconstructs upsampled signals into an original signal.

Abstract: A digital subscriber line communication system does not require the use of a plain old telephone service (POTS) splitter in the resident's home. Digital signal processing techniques are utilized to adapt to varying subscriber line conditions due to POTS telephone equipment. The digital signal processing techniques eliminate the need for a splitter by reducing susceptibility to distortion due to varying subscriber line characteristics. The digital subscriber line modem characterizes the subscriber line under a variety of conditions. The digital subscriber line modem includes a control circuit which performs rapid retrain operation utilizing line characterization information.

Abstract: This invention reformulates the MMSE criterion to apply to systems in which the desired data to be tracked is the ratio of the data appearing in successive observation intervals. The resulting differential MMSE criterion leads to a number of novel algorithms for adaptive implementation of the MMSE receiver. Applications include equalization for single user systems, and multiuser detection, or interference suppression, for direct sequence CDMA with short spreading sequences (i.e., the period of the spreading sequence equals the symbol interval). The invention also provides blind (i.e., without the requirement of a training symbol sequence for the user of interest) equalization and beamforming (using a receive antenna array) for direct sequence CDMA systems with long spreading sequences (i.e., systems in which the spreading sequences are aperiodic, or have period much larger than the symbol interval). The invention enables recovery of the desired symbol sequence up to an unknown phase.

Abstract: When an input digital signal, having a portion with an associated control value, is connected to a digital filter with adaptive compensation, the digital filter with adaptive compensation selects one of a plurality of coefficient scalers (9501, 9601) as directed by a control bus (990). As the input digital signal propagates through the filter, one of a plurality of next coefficient scalers (9502,9602) are selected by the control bus (990). The digital filter with adaptive compensation interleaves in time the coefficients of multiple conventional filters in order to filter each portion of the digital signal according to its associated control value.

Abstract: A method is provided for initiating communications over a channel between a transmitter and a receiver within a specified spectral range, subject to a mask to be applied at the transmitter so as to attenuate at least one segment of the spectral range. The method includes providing to the receiver a definition of the at least one segment and of a level of attenuation to be applied within the at least one segment during a training interval of the communications. A training signal is transmitted from the transmitter to the receiver while applying to the training signal the level of attenuation within the at least one segment. The receiver compensates for a response of the channel based on the received training signal and on the definition of the at least one segment and of the level of attenuation.

Abstract: The present invention provides an optimal procedure for determining in the time domain equalizer coefficients for an equalizer, where the equalizer compensates for the effects of the communications channel on the transmitted signal. A unit pulse or other training sequence which allows determination of the impulse response is transmitted over the communications channel, and a channel impulse response is estimated from the received signal. A cost function establishes a mean-square error associated with the estimated channel impulse response as compared to a desired impulse response signal. The value of the cost function varies based upon an offset value between the estimated channel impulse response and an equalized channel impulse response. Values of the cost function are determined for different offsets, and the offset that produces the smallest cost function value (corresponding to the minimum mean-square error) is selected.

Abstract: Four (4) unshielded twisted pairs of wires connect a hub and a computer in an Ethernet system: one (1) pair for transmission only, another for reception only and the other two (2) for transmission and reception. The signals in the wires are in packets each having timing signals defining a preamble and thereafter having digital signals representing information as by individual ones of three (3) amplitude levels. The signals received at the computer are provided with an automatic gain control (AGC) and then with digital conversions at a particular rate. A control loop operative upon the digital conversions regulates the AGC gain at a particular value. An equalizer operative only during the occurrence of the digital signals in each packet selects an individual one of the three (3) amplitude levels closest to the amplitude of each digital conversion at the time assumed to constitute the conversion peak.