A signal conversion from an input signal to an output signal where the filter used is factorized so that the conversion comprises determining 1) only a first factor at each sampling time of the input signal, where this first factor is independent on the sampling times of the output signal, and 2) only a second factor at each sampling time of the output signal, where this second factor is independent of the sampling times of the input signal. This reduces the computational load for this conversion. In addition, for most filters, the factors may be calculated recursively further increasing the computational load and also reducing the storage requirements. This allows for instantaneous changes in the sampling rates or non-uniform sampling rates with low computational requirements and low memory usage.

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 signal conversion from an input signal to an output signal where the filter used is factorized so that the conversion comprises determining 1) only a first factor at each sampling time of the input signal, where this first factor is independent on the sampling times of the output signal, and 2) only a second factor at each sampling time of the output signal, where this second factor is independent of the sampling times of the input signal. This reduces the computational load for this conversion. In addition, for most filters, the factors may be calculated recursively further increasing the computational load and also reducing the storage requirements. This allows for instantaneous changes in the sampling rates or non-uniform sampling rates with low computational requirements and low memory usage.

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.

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.

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 for resampling an audio-frequency signal with an output sampling frequency, for a current signal frame. The method is used when the preceding frame is sampled at a first sampling frequency which is different from a second sampling frequency of the current frame. The method includes: determining a first and second segments of the signal by adding samples at zero at the end of stored samples of the preceding frame and at the start of samples of the current frame, respectively; obtaining the first resampled segment and the second resampled segment by applying at least one resampling filter respectively to the first segment resampling the first frequency at the output frequency, and to the second segment resampling the second frequency at the output frequency; and combining the overlapping portion of the first and second resampled segments to obtain at least one portion of the resampled current frame.

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.