Apparatus and associated methods relate to an ADC-based digital receiver including a feedforward equalizer (FFE) that has m precursor taps and n postcursor taps to equalize the precursor portion, and to adapt postcursor intersymbol interference (ISI) through a predetermined equalization coefficient selected to counteract the noise boosting effect associated with the precursor equalization. In an illustrative example, the receiver may dynamically balance noise and ISI through adaptively determining a coefficient hp.sub.1 of a first postcursor tap of a first FFE and a coefficient h.sub.1 of a first postcursor tap of a second equalizer adapted to substantially reduce or eliminate additional ISI introduced by the first FFE. The first FFE may optimize ISI removal and noise reduction, for example. One of the coefficients h.sub.1 and hp.sub.1 may be predetermined, and then the other coefficient may be iteratively adapted to trade off precursor ISI and postcursor ISI to minimize BER.

From a mixed signal in which a first signal and a second signal are mixed, the second signal is removed at low processing cost and without delay. As a result, an estimated first signal which has low residue of the second signal and low distortion is obtained. An estimated first signal is generated by subtracting a pseudo second signal which is estimated to be mixed in a first mixed signal in which a first signal and a second signal are mixed from the first mixed signal. The pseudo second signal is obtained by a first adaptive filter using a second mixed signal in which the first signal and the second signal are mixed in a different proportion from the first mixed signal. A coefficient update amount of the first adaptive filter is made smaller as compared with a case when the estimated first signal is smaller than the first mixed signal, in case the estimated first signal is larger than the first mixed signal.

The present invention is a computationally-efficient compensator for removing nonlinear distortion. The compensator operates in a digital post-compensation configuration for linearization of devices or systems such as analog-to-digital converters and RF receiver electronics. The compensator also operates in a digital pre-compensation configuration for linearization of devices or systems such as digital-to-analog converters, RF power amplifiers, and RF transmitter electronics. The adaptive Volterra compensator effectively removes nonlinear distortion in these systems by implementing an adaptive background algorithm to periodically update actual filter coefficients to maintain optimal performance in operating conditions varying over time (e.g., temperature, frequency, signal level, and drift); or both. The xadaptive background algorithm calculates the optimal nonlinear filter coefficients to reduce nonlinear distortion.

A compensator includes a processor, and a storage device that is connected to the processor and stores measured input values that are measured values of an input to a subject to control and measured output values that are measured values of an output from the subject to control. The processor performs processing for identifying, from the measured input values and the measured output values, a first coefficient configuring a finite impulse response filter, processing for determining, based on the first coefficient, a second coefficient configuring an infinite impulse response filter, and identifying an inverse system for the subject to control, the inverse system including the second coefficient and a part of the first coefficient, and processing for estimating an input value to be input to the subject to control from a weighted sum of target values of the output.

A method for adaptive signal processing is provided. In the method, a second vector is obtained by initializing a first vector without regularization of a cost function. The cost function is regularized with the first vector and the second vector as variables. The first vector is updated based on an input signal, according to the regularized cost function. Then, an output signal is provided based on the updated first vector. The second vector is updated based on the update of the first vector. An apparatus for adaptive signal processing is provided accordingly. The method and the apparatus are well compatible with existing adaptive signal processing. The convergence coefficients of the adaptive filter system become more stable. Moreover, impact of an extra penalty added to the cost function on a bias can be minimized, and the increased complexity of the system is very limited.

The present invention is a computationally-efficient compensator for removing nonlinear distortion. The compensator operates in a digital post-compensation configuration for linearization of devices or systems such as analog-to-digital converters and RF receiver electronics. The compensator also operates in a digital pre-compensation configuration for linearization of devices or systems such as digital-to-analog converters, RF power amplifiers, and RF transmitter electronics. The multi-dimensional compensator effectively removes linear and nonlinear distortion in these systems by accurately modeling the state of the device by tracking multiple functions of the input, including but not limited to present signal value, delay function, derivative function (including higher order derivatives), integral function (including higher order integrals), signal statistics (mean, median, standard deviation, variance), covariance function, power calculation function (RMS or peak), or polynomial functions. The multi-dimensional compensator can be adaptively calibrated using simple arithmetic operations that can be completed with low processing requirements and quickly to track parameters that rapidly change over time, temperature, power level such as in frequency-hopping systems.

A method for adaptive signal processing is provided. In the method, a second vector is obtained by initializing a first vector without regularization of a cost function. The cost function is regularized with the first vector and the second vector as variables. The first vector is updated based on an input signal, according to the regularized cost function. Then, an output signal is provided based on the updated first vector. The second vector is updated based on the update of the first vector. An apparatus for adaptive signal processing is provided accordingly. The method and the apparatus are well compatible with existing adaptive signal processing. The convergence coefficients of the adaptive filter system become more stable. Moreover, impact of an extra penalty added to the cost function on a bias can be minimized, and the increased complexity of the system is very limited.

A partitioned block frequency domain adaptive filter device includes a frequency domain adaptive filter configured for filtering a frequency domain representation of a time domain input signal depending on a set of filter coefficients consisting of a plurality of blocks of filter coefficients in order to produce a filtered signal; a plurality of parallel arranged filter update blocks; wherein each of the filter update blocks includes an adaptation module configured for executing an adaptation sequence including the steps of calculating an approximation of a constrained gradient update for the filter coefficients of the respective block of filter coefficients, and calculating a cumulative error introduced on the unconstrained gradient update; wherein each of the filter update blocks includes a correction module configured for executing a correction sequence including the steps of calculating a corrected constrained gradient update for the filter coefficients of the respective block of filter coefficients.

A method for compensating the bandedge ripple of an analog filter, using a circuit comprising a low pass filter is described. The method comprises receiving, at the analog filter, a plurality of tones of different frequencies from a tone generator, measuring, an amplitude of each tone in the plurality of tones after each tone is processed by the analog filter, storing the measured amplitudes and frequencies in a database, measuring a bandedge ripple by measuring a difference in amplitude between a first tone and a second tone from the plurality of tones, and selecting a low pass filter, from a plurality of low pass filters, based on the measured difference.

A partitioned block frequency domain adaptive filter device includes a frequency domain adaptive filter configured for filtering a frequency domain representation of a time domain input signal depending on a set of filter coefficients consisting of a plurality of blocks of filter coefficients in order to produce a filtered signal; a plurality of parallel arranged filter update blocks; wherein each of the filter update blocks includes an adaptation module configured for executing an adaptation sequence including the steps of calculating an approximation of a constrained gradient update for the filter coefficients of the respective block of filter coefficients, and calculating a cumulative error introduced on the unconstrained gradient update; wherein each of the filter update blocks includes a correction module configured for executing a correction sequence including the steps of calculating a corrected constrained gradient update for the filter coefficients of the respective block of filter coefficients.