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.

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.

In order to enhance the speed of a processing necessary for setting the factor of a filter and further maintain the accuracy of the filter, a digital processing apparatus includes: a Fourier transform unit that Fourier transforms a time domain digital signal, thereby generating N frequency domain signals; a filter unit that uses N first factors to process the frequency domain signals in the frequency domain; an inverse Fourier transform unit that transforms the frequency domain signals as processed by the filer unit to a time domain digital signal; a low accuracy factor calculation unit that uses m second factors to calculate N first A factors; a high accuracy factor calculation unit that includes a factor division unit for calculating respective ratios of N third factors to the N first A factors and that also includes a factor variable unit for calculating N first B factors varying stepwise from one to the respective ratios; a multiplication unit that multiplies the first A factors by the first B factors, thereby calculating the N first factors; and a control unit that controls the low accuracy factor calculation unit and high accuracy factor calculation unit by causing only the low accuracy factor calculation unit to operate with the first B factors being set to one and thereafter causing the high accuracy factor calculation unit to calculate the first B factors based on the third factors.

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.

A butterfly channelizer includes at least two stages. Each stage includes at least one dual-channel module configured to convert an input time domain signal into a second time domain signal of lower bandwidth. At least one clock is configured to generate a clock signal that drives the at least two stages. A first stage has a first number of dual-channel modules and a second stage following the first stage has a second number of dual-channel modules greater than the first number.

Reduction of a circuit size and power consumption for performing digital filtering processing in a frequency domain is realized. The digital filter circuit includes: a complex conjugate generation unit for generating a second complex number signal including conjugate complex numbers of all complex numbers included in a first complex number signal of the frequency domain generated by converting a complex number signal of a time domain by Fourier transform; a filter coefficient generation unit for generating a first and a second frequency domain filter coefficient of a complex number from a first, a second and a third input filter coefficient of a complex number having been inputted; a first filtering unit for performing filtering processing to the first complex number signal by the first frequency domain filter coefficient, and outputting a third complex number signal; a second filtering unit for performing filtering processing to the second complex number signal by the second frequency domain filter coefficient, and outputting a fourth complex number signal; and a complex conjugate combining unit for combining the third complex number signal and the fourth complex number signal, and generating a fifth complex number signal.

A filter coefficient group computation device is configured from a means for performing inverse Fourier transform on a frequency characteristic inputted through an input means; a means for performing short-term Fourier transform on a numerical string obtained by the inverse Fourier transform; a means for performing windowing on a frequency domain signal, obtained by the short-term Fourier transform, using a function of which a window length shortens as frequency increases; a means for performing short-term inverse Fourier transform on the frequency domain signal after the windowing; a means for performing overlap addition on a numerical string obtained by the short-term inverse Fourier transform; and a means for determining a numerical string after the overlap addition as a filter coefficient group which forms a filter having the frequency characteristic inputted through the input means.

Methods and systems for resampling an input signal that includes a first plurality of values. A first resampling is performed to obtain an intermediate signal, which includes dividing the first plurality of values into a plurality of groups of values, resampling values in each group according to a first filter tap set to obtain an intermediate group of values. The first value of each group is aligned in time with the first value of each intermediate group. A second resampling is then performed on the intermediate signal to obtain an output signal, which performs a phase shift of each of the intermediate groups of values to align in time with a respective group of values of the output signal. The output signal is output by wired or wireless means.

Fractional Fourier Transform (FrFT)-based spectrum analyzers and spectrum analysis techniques are disclosed. Rather than using the standard Fast Fourier Transform (FFT), the FrFT may be used to view the signal content contained in a particular bandwidth. Usage of the FrFT in place of the frequency or time domain allows viewing of the signal in different dimensions, where spectral features of interest, or signal content, may appear where they were not visible in these domains before. This may allow signals to be identified and viewed in any domain within the continuous time-frequency plane, and may significantly enhance the ability to detect and extract signals that were previously hidden under interference and/or noise, provide or enhance the ability to extract signals from a congested environment, and enable operation in a signal-dense environment.

An apparatus for calibrating a digital filter to replicate a transfer function of a signal processing device comprising a detector with a first input to receive a first signal; a second input to receive a response signal of the signal processing device to the first signal; a signal modification block; a comparison block and a decimation block; wherein the comparison-block compares a phase and amplitude of the first signal after a correction has been applied by the signal modification block and decimation has been applied by the decimation block; and a feedback loop; wherein detector is configured to determine at least a feedback control signal at a first frequency and a second frequency, different to the first frequency and determine calibration information for programming of the transfer function of said digital filter.