In some embodiments, a circuit may include an input configured to receive a signal and a radix-r fast Fourier transform (FFT) processing element coupled to the input. The radix-r FFT processing element may be configured to subdivide data of size N into r equal sub-domains of size N/r1 to determine specific frequencies.

A signal filtering method for filtering a numerical input signal that is sampled at a sampling frequency in order to obtain a filtered signal, the method including a first operation of application of a discrete Fourier transform to M points on a processed signal in order to obtain M points of the spectrum of the processed signal, each point of the spectrum of the processed signal corresponding to the even-numbered indices of a spectral analysis at 2*M points of the processed signal; and a second operation of application of a discrete Fourier transform to M points on the processed signal in order to obtain M points of the spectrum of the processed signal, each point of the spectrum of the processed signal corresponding to the odd-numbered indices of a spectral analysis at 2*M points of the processed signal.

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 radio communication receiver and a method performed by the radio communication receiver for configuring a Notch filter of the radio communication receiver. The method comprises retrieving stored and previously determined filter coefficients from a set of filter coefficients, where the retrieved filter coefficients constitute a fraction of the total number of filter coefficients; and setting the rest of the filter coefficients to one. The method further comprises normalising the retrieved filter coefficients; and transforming the filter coefficients such that the Notch position ends up at the one or more frequencies to be filtered out.

A memory device includes a memory configured to store input data and filter data for a convolution operation, and a function processor configured to, in response to a read command of at least a portion of data from among the input data and the filter data, transform the at least a portion of the data based on a parameter of the convolution operation during a clock cycle corresponding to the read command and output a corresponding transformation result as transformed data.

A device for processing a digital signal includes a Farrow structure (14) that applies to the digital signal a time-varying sample rate conversion from the fixed sample rate to a time varying sampling. The digital signal sampled at the time varying sampling is a resulting signal. The Farrow structure (14) is controlled from a control variable. A spectral analysis means (15) performs a spectral analysis of the resulting signal to determine the frequency values of the resulting signal. A determining means (16) determines a sparseness parameter of the frequency values of the resulting signal. A controlling means (17) modifies the control variable according to the value of the sparseness parameter.

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 SOI estimate and the true SOI is less than or equal to an error threshold (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 algorithmsindeed, by an order of magnitude or more.

An apparatus for modifying a sampling rate includes a forward transformer for forming a first version of a spectrogram by means of transformation with a first transformation length from an information signal with a first sampling rate. The apparatus includes a processor for forming a second version of the spectrogram with a lower bandwidth than the first version. The apparatus includes a reverse transformer for forming a coarsely pre-modified information signal with a second sampling rate that is reduced with respect to the first sampling rate, by means of reverse transformation of the second version of the spectrogram with a second transformation length that is reduced with respect to the first transformation length. The apparatus includes a time domain interpolator for acquiring an information signal with a third sampling rate that is modified with respect to the second sampling rate, by means of interpolation of the pre-modified information signal.

A spot noise generator includes a mask component, a polyphase synthesizer, a first signal channel and second signal channel. The mask component has a narrowband noise input, a desired frequency channels word input, a first channel output and a second channel output. The narrowband noise input signal is a digital narrowband noise signal sampled approximately at the Nyquist rate. The desired frequency channels word selects one of the group consisting of the first channel output, the second channel output and a combination of the first channel output and the second channel output. The polyphase synthesizer synthesizes the first channel output signal, synthesizes the second channel output signal and outputs a desired noise signal based on the synthesized first channel output signal and the synthesized second channel output signal.

An audio processor (1) includes a first filter coefficient calculator (31) that calculates a first filter coefficient so as to correspond to first gains for respective bands set by a user, a second filter coefficient calculator (32) that if values of third gains for respective bands of the first filter coefficient are greater than an absolute value of a second gain set by the user, calculates a second filter coefficient by limiting the values of the third gains for the respective bands to the amplitude value of the second gain, and a filtering unit (35) that filters an audio signal that has been transformed into a frequency-domain signal, using the second filter coefficient.