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.

Provided is a digital filter implementing a function by using circular convolution of a digital input signal and a unit pulse response of the digital filter in applying a filter to the digital signal. In addition, a configuration sequence change of an output signal, which occurs as a result of digital filtering according to the present invention, in comparison to an input signal, is corrected by inversely performing circular shifting in a transmitter by amount of the configuration sequence change, or by reconstructing a signal after making synchronization and acquiring the signal previous to a time point of the synchronization by amount of the configuration sequence change, in a receiver.

Disclosed herein are system, method, and computer program product embodiments for synchronizing the switching of different wireless platforms to different portions of a frequency band. An embodiment, at a first wireless platform, operates by receiving a band switch request message from a second wireless platform, wherein the band switch request message comprises a band switch delay period for the second wireless platform. The embodiment calculates a band switch time based on a band switch delay period for the first wireless platform and the band switch delay period for the second wireless platform. The embodiment transmits a band switch accept message comprising the band switch time to the second wireless platform. The embodiment sets a first filter to operate on a second portion of the frequency band based on the band switch time. The embodiment then operates on the second portion of the frequency band.

A method for acoustic source separation comprises inputting acoustic data from a plurality of acoustic sensors, combined from a plurality of acoustic sources, converting the acoustic data to time-frequency domain data comprising time-frequency data frames, and constructing a multichannel filter for the time-frequency data frames to separate signals from the acoustic sources. The constructing comprises determining a set of de-mixing matrices (W.sub.f) to apply to each time-frequency data frame to determine a vector of separated outputs (y.sub.ft) by modifying each of the de-mixing matrices by a respective gradient value (G;G′) for a frequency dependent upon a gradient of a cost function measuring a separation of the sources by the respective de-mixing matrix. The respective gradient values for each frequency are each calculated from a stochastic selection of the time-frequency data frames.

A system performs pole-zero or IIR modeling and estimation of an inter-microphone transfer function between first and second microphones that output respective first and second microphone signals. The system includes a first adaptive FIR filter to which the first microphone signal is provided, a delay element that delays the second microphone signal by a predetermined delay amount, and a second adaptive FIR filter to which the delayed second microphone signal is provided. A first coefficient of the second adaptive FIR filter is constrained to a fixed non-zero value. The filters are jointly adapted to minimize an error signal that is a difference of the two filters outputs. The delay is small: approximately the acoustic propagation delay between the two microphones and is not determined by the environmental reverberation characteristics. The error signal may serve as a noise reference in a noise canceller, for implementing far-field beamforming with low delay.

A system performs pole-zero or IIR modeling and estimation of an inter-microphone transfer function between first and second microphones that output respective first and second microphone signals. The system includes a first adaptive FIR filter to which the first microphone signal is provided, a delay element that delays the second microphone signal by a predetermined delay amount, and a second adaptive FIR filter to which the delayed second microphone signal is provided. A first coefficient of the second adaptive FIR filter is constrained to a fixed non-zero value. The filters are jointly adapted to minimize an error signal that is a difference of the two filters outputs. The delay is small: approximately the acoustic propagation delay between the two microphones and is not determined by the environmental reverberation characteristics. The error signal may serve as a noise reference in a noise canceller, for implementing far-field beamforming with low delay.

Disclosed herein are system, method, and computer program product embodiments for synchronizing the switching of different wireless platforms to different portions of a frequency band. An embodiment, at a first wireless platform, operates by receiving a band switch request message from a second wireless platform, wherein the band switch request message comprises a band switch delay period for the second wireless platform. The embodiment calculates a band switch time based on a band switch delay period for the first wireless platform and the band switch delay period for the second wireless platform. The embodiment transmits a band switch accept message comprising the band switch time to the second wireless platform. The embodiment sets a first filter to operate on a second portion of the frequency band based on the band switch time. The embodiment then operates on the second portion of the frequency band.

An adaptive filter calculates frequency domain coefficients and in the frequency domain dynamically adjusts a leakage/step size parameter that controls adaptation of the adaptive filter based on the calculated frequency domain coefficients (e.g., based on a peak magnitude of the coefficients among frequency bins or on the magnitude of the coefficient of the corresponding frequency bin). The adaptive filter calculates the coefficients based on frequency domain input and error signals, dynamically adjusts a frequency domain coefficient magnitude limit parameter based on the calculated frequency domain coefficients (e.g., approximately proportionally to a peak magnitude of the coefficients among frequency bins) and uses the dynamically adjusted frequency domain coefficient magnitude limit parameter to limit a magnitude of the calculated frequency domain coefficients. The limit may be engaged above a frequency bin based on the peak magnitude frequency bin. An ANC system may employ the filter.

A method for acoustic source separation comprises inputting acoustic data from a plurality of acoustic sensors, combined from a plurality of acoustic sources, converting the acoustic data to time-frequency domain data comprising time-frequency data frames, and constructing a multichannel filter for the time-frequency data frames to separate signals from the acoustic sources. The constructing comprises determining a set of de-mixing matrices (W.sub.f) to apply to each time-frequency data frame to determine a vector of separated outputs (y.sub.ft) by modifying each of the de-mixing matrices by a respective gradient value (G;G) for a frequency dependent upon a gradient of a cost function measuring a separation of the sources by the respective de-mixing matrix. The respective gradient values for each frequency are each calculated from a stochastic selection of the time-frequency data frames.

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.