A system and method for representing quasi-periodic waveforms, for example, representing a plurality of limited decompositions of the quasi-periodic waveform. Each decomposition includes a first and second amplitude value and at least one time value. In some embodiments, each of the decompositions is phase adjusted such that the arithmetic sum of the plurality of limited decompositions reconstructs the quasi-periodic waveform. Data-structure attributes are created and used to reconstruct the quasi-periodic waveform. Features of the quasi-periodic wave are tracked using pattern-recognition techniques. The fundamental rate of the signal (e.g., heartbeat) can vary widely, for example by a factor of 2-3 or more from the lowest to highest frequency. To get quarter-phase representations of a component (e.g., lowest frequency “rate” component) that varies over time (by a factor of two to three) many overlapping filters use bandpass and overlap parameters that allow tracking the component's frequency version on changing quarter-phase basis.

An analysis filter bank corresponding to multiple sub-bands, which performs frequency-division filtering on an input signal to generate multiple sub-band signals, the analysis filter bank comprising: a sub-band response pre-compensator which performs a linear filtering on the input signal to generate a response pre-compensated signal, multiple sub-filters with different central frequencies, which perform complex-type first-order infinite impulse response filtering respectively on the response pre-compensated signal to generate multiple sub-filter signals, and multiple binomially-combining and rotating devices based on a set of binomial weights, each of which performs a weighted summation on at least two of the sub-filter signals with the set of binomial weights, and rotates a weighted-summation result with a rotating phase according to a corresponding sub-band central frequency to generate one of the sub-band signals, wherein the at least two of the sub-filter signals are generated by at least two of the sub-filters adjacent in central frequency.

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.

This application relates to a signal filtering device. The device includes a memory and a processor. The processor may generate one or more matrices based on a size of a digital filter bank that generates an output signal by dividing an input signal into a plurality of channels and store in the memory each of the generated one or more matrices to which a plurality of digital filter bank coefficients or a plurality of input signals are assigned. The processor may also partially calculate the stored plurality of digital filter bank coefficients and the plurality of signals based on a number of at least some of the plurality of channels, and calculate the calculated digital filter bank coefficients and signals. The processor may further perform a discrete Fourier transform (DFT) on the calculated signal and compensate for a phase of the discrete Fourier transformed signal according to a preset reference.

The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

An analysis filter bank corresponding to multiple sub-bands, which performs frequency-division filtering on an input signal to generate multiple sub-band signals, the analysis filter bank comprising: a sub-band response pre-compensator which performs a linear filtering on the input signal to generate a response pre-compensated signal, multiple sub-filters with different central frequencies, which perform complex-type first-order infinite impulse response filtering respectively on the response pre-compensated signal to generate multiple sub-filter signals, and multiple binomially-combining and rotating devices based on a set of binomial weights, each of which performs a weighted summation on at least two of the sub-filter signals with the set of binomial weights, and rotates a weighted-summation result with a rotating phase according to a corresponding sub-band central frequency to generate one of the sub-band signals, wherein the at least two of the sub-filter signals are generated by at least two of the sub-filters adjacent in central frequency.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount.

To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.

This application relates to a signal filtering device. The device includes a memory and a processor. The processor may generate one or more matrices based on a size of a digital filter bank that generates an output signal by dividing an input signal into a plurality of channels and store in the memory each of the generated one or more matrices to which a plurality of digital filter bank coefficients or a plurality of input signals are assigned. The processor may also partially calculate the stored plurality of digital filter bank coefficients and the plurality of signals based on a number of at least some of the plurality of channels, and calculate the calculated digital filter bank coefficients and signals. The processor may further perform a discrete Fourier transform (DFT) on the calculated signal and compensate for a phase of the discrete Fourier transformed signal according to a preset reference.

A transmitter module for a broadband data transmission system for radio communications, comprising at least one polyphase FFT filter bank is described. The at least one polyphase FFT filter bank is established as a synthesis polyphase FFT filter bank, wherein the at least one polyphase FFT filter bank comprises several filter units, wherein the transmitter module is configured to receive an input signal comprising a symbol sequence, and wherein the transmitter module is configured to transmit a signal based on the received input signal. Moreover, a receiver module for a broadband data transmission system and a data transmission system are described.

The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.