A signal processing system is described. The signal processing system includes at least one signal processing path and a control module. The at least one signal processing path includes at least one signal input and at least two filter units. The at least two filter units include at least one hardware filter unit. The at least one signal input is connectable to at least one external electronic component. The control module is connected to the signal input and to the at least two hardware filter units. The control module is configured to determine a frequency response deviation being associated with the at least one external electronic component. The control module further is configured to reconfigure the at least one hardware filter unit such that the frequency response deviation is compensated at least partially. Further, a signal processing method for adapting filter coefficients of a signal processing system is described.

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 selectable bandwidth filter has an analysis filter bank and a synthesis filter bank having M paths. A masking vector is disposed between the analysis filter bank and the synthesis filter bank. The masking vector enables select ones and disables unselected ones of the M paths, so as to define an output signal bandwidth.

A filter for use with a brushless DC motor to filter a signal received from a floating terminal of the brushless DC motor, wherein the filter is configured such that a time delay introduced by the filter to the signal received from the floating terminal is equal to the time taken for a rotor of the motor to rotate through an angle equal to half of a commutation step of the motor.

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.

A digital filter circuit is described. The digital filter circuit includes a digital filter input, at least two finite impulse response (FIR) filter circuits, and a connection circuit. The digital filter input is configured to receive a digital input signal set having a data parallelism. The at least two FIR filter circuits are configured to process the digital input signal set at least partially. The at least two FIR filter circuits include a pre-adder sub-circuit, a convolution sub-circuit, and a post-adder sub-circuit, respectively. The connection circuit is configured to selectively connect the at least two FIR filter circuits based on the data parallelism of the digital input signal set.

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.

Digital signal processing (“DSP”) block circuitry on an integrated circuit (“IC”) is adapted for use, e.g., in multiple instances of the DSP block circuitry on the IC, for implementing finite-impulse-response (“FIR”) filters that are dynamically adjustable. Advantages of such DSP block circuitries may include an increase in performance and a reduction in logic and memory usage for multi-standard FIR filters.

A discrete-time (e.g., digital) filter can be used as an interpolation filter for processing an oversampled input signal, such as included as a portion of a sigma-delta digital-to-analog conversion circuit. An interpolation filter control circuit can be configured to adjust a filter order of the discrete-time interpolation filter at least in part in response to information indicative of an envelope signal magnitude. For example, higher-level input signals might be processed using an interpolation filter having a stop-band attenuation that is more stringently-specified (e.g., having greater attenuation) than a corresponding attenuation used for lower-level input signals. The filter order can be variable, such as varied in response to a detected envelope magnitude of the input signal to achieve power savings as compared to a filter having fixed parameters.

The invention relates to a signal processing device, a control method thereof, a control procedure and a recording medium, which is unnecessary to increase a memory capacity for implementing sampling to determine a filter parameter. The signal processing device includes a data acquiring element (10), acquiring signals of a time series, i.e. time series signals from a sensor; a filtering element (21), performing a filtering operation according to frequencies; a transfer element (30), transferring the time series signals; and a filter parameter determination element (50), performing frequency analysis to the time series signals within a pre-specified interval, i.e. a specified interval to determine a filter parameter.