The invention relates to a signal processing device, a signal processing method, an information processing program and a recording medium, which removes wide-range frequency noise. In a signal processing device (10), a cycle of acquiring a signal from a sensor is a data acquisition cycle, which is shorter than a cycle of forwarding time sequence data from which noise is removed to a controller, that is, a forwarding cycle.

A finite impulse response (FIR) filter including a delay line and a plurality of arithmetic units. Each arithmetic unit is coupled to a different one of a plurality of tap points of the delay line, is configured to receive a respective signal value over the delay line, and is associated with a respective coefficient. Any given one of the arithmetic units is configured to receive a respective control word. The respective control word specifying: (i) a plurality of trivial multiplication operations, and (ii) a plurality of bit shift operations. Any given one of the arithmetic units is further configured to estimate or calculate a product of the respective signal of the arithmetic unit respective signal value and the respective coefficient of the arithmetic unit by performing the trivial multiplication operations and bit shift operations that are specified by the respective control word that is received at the given arithmetic unit.

Dual mode filters having two reconfigurable multi-stage filters. In a dual band mode, each reconfigurable filter filters an input signal in a different band using every filter stage. In a single band mode, both reconfigurable filters are effectively divided into two sub-chains that include either the odd-numbered filter stages or the even-numbered filter stages. Together, the four sub-chains in the single band mode filter an input signal in a single band with a higher parallelization than each reconfigurable filter in the dual band mode. In some embodiments, the dual mode filter is a decimation filter. In other embodiments, the dual mode filter is a resampling filter. In still other embodiments, the dual mode filter is an interpolation filter.

The present disclosure relates to a concept of nonlinear signal processing which may be used for predistortion for RF power amplifiers. The concept includes generating time variant filter coefficients for a linear filter circuit based on a nonlinear mapping of an input signal, and filtering the input signal with the linear filter circuit using the time variant filter coefficients in order to generate a filtered output signal. Thus, it is proposed to implement a non-linear filter by a time-varying linear filter where the time-varying coefficients are derived from the input signal.

A finite impulse response (FIR) filter including an input of the FIR filter that receives an RF input signal, a clock input configured to receive a clock signal, an output of the FIR filter that provides a filtered output signal, a plurality of signal paths including a plurality of sample-and-hold circuits and a plurality of multipliers arranged in parallel, each signal path including a respective sample-and-hold circuit and a respective multiplier being configured to receive the RF input signal and the clock signal to provide a modulated output signal, an adder configured to receive n modulated output signals from the plurality of signal paths and combine the n modulated output signals to produce the filtered output signal, and a controller.

Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a subband of the audio signal. The method also includes determining filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

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 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.

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.

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.