Circuit and method for modulating filter coefficients of a frequency channelizer having a filter bank include: receiving a wide spectrum input signal; modulating the filter coefficients of the filter bank to sweep a center frequency of each channel of the frequency channelizer, using a modulation scheme; and inputting frequency offset compensation caused by the modulation, and output signals of the frequency channelizer to an application processing circuit to convert the output signals to their original center frequencies.

An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.

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 audio signal processing apparatus comprises a receiver (403) receiving an audio signal sampled at a first sampling frequency, the audio signal having a maximum frequency below half the first sampling frequency by a first frequency margin. A tilter bank (405) generates subband signals for the digital audio signal using overlapping sub-filters. A first frequency shifter (407) applies a frequency shift to at least one subband of the set of subbands and a decimator (409) decimates the subband signals by a N decimation factor resulting in a decimated sampling frequency being at least twice a bandwidth of each of the overlapping sub-filters. The frequency shift for a subband is arranged to shift the subband to a frequency interval being a multiple of a frequency interval from zero to half the decimated sample frequency. The subband may be individually processed and the processed subbands may subsequently be combined to generate a full band output signal.

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.

The application relates to a filter bank for an audio processing device, e.g. a hearing aid. The filter bank comprises an analysis filter bank comprising a plurality of M first filters h.sub.m(n), where m=0, 1, . . . , M1 is a frequency band index, n being a time index, the first filters h.sub.m(n) having a first filter length of L.sub.hm; a synthesis filter bank comprising a plurality of M second filters g.sub.m(n), m=0, 1, . . . , M1, the second filters g.sub.m(n) having second filter lengths of L.sub.gm; the plurality of first and second filters being arranged in pairs, each pair forming a frequency channel. the first filters h.sub.m(n) exhibiting a first filter delay .sub.h, the second filters g.sub.m(n) exhibiting a second filter delay .sub.g, each of the first filter lengths L.sub.hm and the second filter lengths L.sub.gm is uneven, and wherein the first filters are subject to the constraint that the sum of the first filters h.sub.m(n) of the analysis filter bank is a delta function (n.sub.h).

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 embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.

An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.

An embodiment of an apparatus for generating audio subband values in audio subband channels includes an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function including a sequence of window coefficients to obtain windowed samples. The analysis window function includes a first number of window coefficients derived from a larger window function including a sequence of a larger second number of window coefficients, wherein the window coefficients of the window function are derived by an interpolation of window coefficients of the larger window function. The apparatus further includes a calculator for calculating the audio subband values using the windowed samples.