A signal-of-interest (SOI) may be separated from interference and/or noise using repeated reduced rank minimum mean-square error Fractional Fourier Transform (MMSE-FrFT) filtering and a low rank adaptive multistage Wiener filter (MWF). A number of stages in the MWF, L, may be chosen such that at the L.sup.th stage, the MSE between the SIM estimate and the true SW is less than or equal to an error threshold E (e.g., =0.001). By combining these filtering techniques, significant improvement in reducing the mean-square error (MSE) may be realized over single stage MMSE-FrFT, repeated MMSE-FrFT, and MMSE-FFT algorithms indeed, by an order of magnitude or more.

The equalization filter implements an equalization of at least one signal distorted by a measurement setup. The filter coefficients of the equalization filter can be determined by minimizing a cost function K in which only sequences of filter coefficients which exert significant influence on the equalization are taken into consideration.

A digital signal synthesizer for generating a frequency and/or phase modified digital signal output comprises an input buffer, a transform module, a processing module, and an output buffer. The input buffer receives a digital input that is represented in a frequency domain representation. The transform module stores a fractional order control system that models a desired frequency and/or phase response defined by an assembly of at least one filter component. Each filter component is defined by a Laplace function that is modified to include a non-integer control order having a variable fractional scaling exponent. The processing module multiplies or divides the digital input with the fractional order control system stored in the transform module. Moreover, the output buffer stores a synthesized output of the input, which is modified in the frequency domain, the phase domain, or both according to the desired frequency and/or phase response by the processing module.

The invention relates to a method for filtering a numerical input signal sampled at a sampling frequency in order to obtain a filtered signal, the method being performed by a radar system and including at least one step for obtaining a first (respectively second) output signal by carrying out first (respectively second) operations on the first (respectively second) processing channel, the first (respectively second) operations including at least the application of a discrete Fourier transform to M/2 points on a signal coming from the input signal, and applying an inverse discrete Fourier transform to M/2 points on the first signal in order to obtain M points of the spectrum of the first signal, M being an integer strictly greater than 2, the application step being carried out by the addition of the results of two processing channels.

A parallel transfer rate converter inputs first parallel data with number of samples being S1 pieces in synchronism with a first clock, and outputs second parallel data with number of samples being S2=S1(m/p) pieces (p is an integer equal to or larger than 1) in synchronism with a second clock having a frequency which is p/m times of a frequency of the first clock. A convolution operation device inputs the second parallel data in synchronism with the second clock, generates third parallel data with number of samples being S3=S2(n/m) pieces (S3 is an integer equal to or larger than 1) by executing a convolution operation with a coefficient indicating a transmission characteristic to the second parallel data, and outputs the third parallel data in synchronism with the second clock.

A signal processing arrangement generates electrical stimulation signals to electrode contacts in an implanted cochlear implant array. An input sound signal is processed to generate band pass signals that each represent an associated band of audio frequencies. A spectrogram representative of frequency spectrum present in the input sound signal is generated. A characteristic envelope signal is produced for each band pass signal based on its amplitude. An active contour model is applied to estimate dominant frequencies present in the spectrogram, and the estimate is used to generate stimulation timing signals for the input sound signal. The electrode stimulation signals are produced for each electrode contact based on the envelope signals and the stimulation timing signals.

The present invention is aimed at providing a switching power supply apparatus enabling to avoid enhancement of noise of a specific frequency component. The present invention is a switching power supply apparatus provided with: a switching power supply configured to perform switching of input from a primary power source and thereby output a secondary power source; and a noise frequency analysis device configured to analyze frequency components of noise included in output of the primary power source or of the secondary power source, and accordingly cause the switching power supply to perform switching at a different frequency from a frequency of a maximum noise amplitude among the frequency components.

Disclosed herein, among other things, are systems and methods for reducing the comb filtering for hearing devices. A method includes determining an effective time delay as a function of frequency between a direct path from a sound source to an eardrum of a wearer of a hearing device and an amplified path from the sound source through the hearing device to the eardrum of the wearer, and calculating a set of cancellation frequencies based at least in part on the determined effective time delay. The method also includes comparing an aided gain response and a real-ear occluded gain response of the hearing device, and determining one or more spectral interaction regions based at least in part on the comparison. A subset of cancellation frequencies is determined by comparing the spectral interaction regions with the set of cancellation frequencies, and a filter is constructed using the subset of cancellation frequencies and a determined attenuation configured to reduce comb filtering.

A signal processing apparatus has a first memory in which plural pieces of FIR coefficient data used for implementing an FIR filter algorithm are stored, a second memory which stores plural pieces of input data to be subjected to the FIR filter algorithm, and a processor implements the FIR filter algorithm using the plural pieces of FIR coefficient data stored in the first memory and the plural pieces of input data stored in the second memory as many times as the number corresponding to a designated filter order, in which filter algorithm each piece of coefficient data and each piece of input data are multiplied together and resultant products are summed up. The signal processing apparatus is provided, which can implement plural sorts of FIR filter algorithms of filter order which can be changed flexibly.

A filter generator (100) generates a filter on the basis of band information (frequency) and gain characteristics (gain value) set by a user. The filter generator (100) obtains weighting factor information on the basis of the band information selected by the user and calculates a gain difference between a gain value used in a preceding filtering process and the new gain value selected by the user. The filter generator (100) then obtains a correction gain by multiplying the weighting factor information by the gain difference and generates a filter by multiplying a coefficient of the filter used in the preceding filtering process by the correction gain.