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.

The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(f.sub.n, A(t.sub.1, . . . , t.sub.M(fn))) in the frequency domain; to calculate a transformed second audio signal Y(f.sub.n, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal.

The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(f.sub.n, A(t.sub.1, . . . , t.sub.M(fn))) in the frequency domain; to calculate a transformed second audio signal Y(f.sub.n, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal.

A polyphase decimation FIR filter apparatus including a modulo integrator circuit configured to integrate input samples and to provide integrated input samples; and a polyphase FIR filter circuit configured to process the integrated input samples, the polyphase FIR filter circuit including a plurality of multiplier accumulator circuits, each configured to accumulate products of coefficients and respective integrated signal samples, wherein each of the multiplier accumulator circuits receives a subset of FIR filter coefficients, wherein the FIR filter coefficients are derived as the nth difference of original filter coefficients, where n is a number of integrators in the integrator circuit, and wherein the FIR filter circuit is configured to perform computation operations with modulo arithmetic.

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.

The invention is a method that combines the following components: a high pass filter designed to have sufficiently small phase delay and roll-off value in transition band as well as sufficiently good attenuation; a distortion detection and reconstruction introduced by the application of the high pass filter by extraction the significant frequency components in relevant frequency band; a signal compensation that reshapes the output of the high pass filter by matching the filter's phase delay and attenuation characteristics so as to approximate low frequency component extraction that would be produced by an ideal filter (very sharp frequency transition and no delay); a time-domain detection and correction method that addresses special circumstances under which the compensation would be inaccurate to achieve real-time estimate in normal circumstances, and a time-domain correction method during and immediately after sudden changes in composite signal (spike) is detected.

The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(f.sub.n, A(t.sub.1, . . . , t.sub.M(fn))) in the frequency domain; to calculate a transformed second audio signal Y(f.sub.n, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal.

The invention is a method that combines the following components: 1. a high pass filter designed to have sufficiently small phase delay and roll-off value in transition band as well as sufficiently good attenuation; 2. a distortion detection and reconstruction introduced by the application of the high pass filter by extraction the significant frequency components in relevant frequency band; 3. a signal compensation that reshapes the output of the high pass filter by matching the filter's phase delay and attenuation characteristics so as to approximate low frequency component extraction that would be produced by an ideal filter (very sharp frequency transition and no delay); 4. a time-domain detection and correction method that addresses special circumstances under which the compensation would be inaccurate to achieve real-time estimate in normal circumstances, and 5. a time-domain correction method during and immediately after sudden changes in composite signal (spike) is detected.

The systems and methods described provide a solution for automatic training of echo canceller (EC) in upstream signal receivers on sub-carriers experiencing low activity in the upstream communication. The solution can include one or more processors identifying a plurality of bins (e.g., EC bins). Each bin of the plurality of bins can correspond to a portion of a frequency range of a cable line. The one or more processors can determine that a power level of an echo signal of each of a bin of the plurality of bins and one or more bins adjacent to the bin exceed a respective threshold for the power level. The one or more processors can train an echo canceller of the echo signal for the bin, responsive to the determining.