An example process includes reducing a quality factor of a first tunable bandpass filter, used, for example, in a low-noise amplifier stage of a polar receiver. A first wideband test signal centered at a desired center frequency of a second tunable bandpass filter is received. A frequency response of the second tunable bandpass filter to the first wideband test signal is estimated using a Fast Fourier Transform (FFT) signal processor. At least a resonant frequency or a quality factor of the second tunable bandpass filter are calibrated based at least in part on a portion of the estimated frequency response of the second tunable bandpass filter obtained from the FFT signal processor. Frequency response characteristics of the first tunable bandpass filter may be similarly tuned in accordance with the example process.

An equalizer has a first tapped delay line in which N taps (N is a positive integer) are connected in cascade, a second tapped delay line having one tap and connected in parallel with the first tapped delay line, a first multiplier configured to multiply signals extracted from the N taps by corresponding coefficients, a second multiplier configured to multiply a signal output from the second tapped delay line by a second coefficient, and an adder configured to add products of the first multiplier and a product of the second multiplier. The first tapped delay line has a fixed delay, and the second tapped delay line has a variable delay changeable at a 1/M resolution of the fixed delay, where M is a number greater than 1.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

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.

An optical communication apparatus includes a level detector, an FIR filter, and a adjustor. The level detector detects level information that discriminates a change in a multi-value level based on an input signal used in a multi-value amplitude modulation system. The FIR filter compensates a signal band of the input signal in accordance with tap coefficients of a plurality of multipliers. The adjustor corrects the tap coefficient of each of the multipliers included in the FIR filter based on the level information detected in the level detector.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

A digital interpolation filter delivering a series of output samples approximating a signal x(t) at sampling instants of the form (n+d)T s based on a series of input samples of the signal x(t) taken at sampling instants of the form nT s. Such a filter implements a transfer function in the Z-transform domain, H c<i/>d (Z−1), expressed as a linear combination between: a first transfer function H 1 d<i/>(Z−1) representing a Lagrange polynomial interpolation of the input samples implemented according to a Newton structure (100); and a second transfer function H 2 d (Z−1) representing another polynomial interpolation of the input samples implemented according to another structure comprising at least the Newton structure; the linear combination being a function of at least one real combination parameter c.

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 Finite Impulse Response (FIR) filter is configured to minimize delay and maximize passband power by adjusting the filter coefficients applied to the sampled values. The FIR filter obtains an input signal and samples the input signal to generate a set of sampled input values. The FIR filter generates a set of filter coefficients, with each filter coefficient based on a corresponding sampled input value in the set of sample input values. The FIR filter selects a subset of sampled input values that have been most recently sampled from the input signal, and selects a subset of filter coefficients corresponding to sampled input values that are not the most recently sampled. The subset of sampled input values is combined with the subset of filter coefficients to generate an output value for the FIR filter.