One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

An integrated circuit device includes a digitally controlled oscillator (DCO), two charge-sharing capacitors, two charge-sharing switches, two pre-charge switches, and two DACs. The DCO has a first inverter and a second inverter. A first charge-sharing capacitor has a first terminal coupled to an input terminal of the first inverter through a first charge-sharing switch. A first DAC has an output terminal coupled to the first terminal of the first charge-sharing capacitor through a first pre-charge switch. A second charge-sharing capacitor has a first terminal coupled to an input terminal or an output terminal of the second inverter through a second charge-sharing switch. A second DAC has an output terminal coupled to the first terminal of the second charge-sharing capacitor through a second pre-charge switch.

The embodiments of the present disclosure provide a method and system for adjusting a passband width of a filter. The method includes determining an initial passband width, controlling a filter according to the initial passband width to filter signals to be processed, and correcting the initial passband width according to a first frequency at which a peak spectrum line corresponding to the filtered signals is located.

One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

One example includes a digital signal conditioner (DSC) system. A sample selector bank receives a digital sample block of an input signal that is provided at a supported input oversampling factor and selects a subset of samples from the digital sample block based on a selection signal. A tap weights selector bank generates a set of tap weights based on the selection signal. A filter bank receives the subset of the samples from each of the sample selectors and a respective set of tap weights. Each filter provides a weighted sample associated with the respective subset of samples and the respective set of tap weights. A reformattor receives the weighted sample from each of the filters and provides a filtered sample block including the weighted sample from a subset of the filters at an output oversampling factor for each supported input oversampling factor based on a selected supported resampling ratio.

A Scalable Finite Impulse Response (SFIR) filter is disclosed. The SFIR filter includes a pre-processing section, a post-processing section, and a finite impulse response (FIR) Matrix. The FIR Matrix includes a plurality of filter taps and a plurality of signal paths in signal communication with each filter tap. The plurality of signal paths are arranged to allow re-configurable data throughput between the each filter tap and the pre-processing section and post-processing section are in signal communication with the FIR Matrix.

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

The present invention relates broadly to a method of digitally filtering a signal, such as an audio signal, using a digital filter. The digital filter includes a plurality of neighbouring sample points broken into zones having different frequency content or frequency ranges. The zones adjacent one another may have neighbouring sample points in common. Generally each zone has at least same distinct frequencies compared with other zones. That is, the zones are roughly dependent on the frequency content. The invention in its preferred form involves combining values for two or more of the neighbouring sample points for select of the zones depending on its frequency content. The values are combined so as to provide a modified zone having substantially the same number of sample points as the select zone. The modified zones together provide a modified filter to be applied to the signal.

A method in an adaptive filter system is provided. The method comprises obtaining parameters for a plurality of branches of the adaptive filter system. The method further comprises computing gradient-based information for a selected one of the plurality of branches. The method further comprises updating the parameters for the plurality of branches based on the gradient-based information for the selected branch. An adaptive filter system is also provided.

The present invention discloses a method for determining FIR digital filter coefficients providing a compensated amplitude response of a conventional windowing filter. The compensation allows independent control of passband and stopband specifications, while narrowing the transition bandwidths. The method comprises the steps of summing an auxiliary impulse response to a windowing impulse response, wherein the auxiliary impulse response has a phase response identical to that of the windowing impulse response, and an amplitude response comprising frequency shifted pulses positioned to induce the required compensation. The summing is followed by a modulation with a discrete time function to obtain the filter impulse response. The invention discloses a characterization of the pulse compensation on the amplitude response, a filter design method and computer program based on this characterization. The invention discloses an FIR filter device with coefficients set by this method.