Electrical audio signal processing systems and devices – Hearing aids – electrical – Frequency transposition
Reexamination Certificate
1998-04-16
2001-05-22
Kuntz, Curtis (Department: 2643)
Electrical audio signal processing systems and devices
Hearing aids, electrical
Frequency transposition
C381S074000, C381S312000, C381S314000
Reexamination Certificate
active
06236731
ABSTRACT:
FIELD OF THE INVENTION
This invention relates to a filterbank structure and a method for filtering and separating an information signal into different bands, particularly for such filtering and separation of audio signals in hearing aids. This invention more particularly relates to such a technique carried out using digital signal processing in hearing aids. This invention more particularly relates to a method and architecture for a digital filterbank for hearing aid applications.
BACKGROUND OF THE INVENTION
Hearing loss is generally associated with a loss of hearing sensitivity which is a function of frequency. The most common type of sensitivity loss is an increasing function of frequency. Sensitivity is typically a function of speech level as well. Hence, loud sounds should be amplified less than soft sounds. It has been long known that a hearing aid should treat the various frequency components of speech differently to render them intelligible to a hearing impaired person.
Known analog hearing aids use relatively simple methods to alter their frequency shaping and dynamic range compression to mitigate the loss in hearing sensitivity for frequency and level.
SUMMARY OF THE INVENTION
Digital techniques promise far greater possibilities for signal processing to aid the hearing impaired. The present inventors have combined processing. This allows great processing flexibility as the bands can be treated independently to compensate more precisely for hearing loss.
In accordance with the first aspect of the present invention, there is provided an oversampled filterbank for filtering an information signal, the filterbank having a filterbank structure comprising a filter means defining a filter bandwidth, said filter means filtering said information signal and separating said information signal into a plurality of frequency band signals each representing one of a plurality of uniformly spaced frequency bands within said filter bandwidth, said frequency bands being stacked in one of an even and an odd manner and said frequency bands overlapping, such that the summation of the unmodified frequency hand responses of the plurality of said frequency bands sums to a function within a predetermined passband ripple over said filter bandwidth, wherein the filter means includes a selection input enabling at least one of the following to be selected:
(i) the number of frequency band signals,
(ii) the bandwidth of said frequency bands,
(iii) selection of stacking of said frequency bands in one of an even and an odd manner,
(iv) the degree of overlap between said frequency bands,
(v) an oversampling factor by which said frequency band signals are sampled above the theoretical minimum of critical sampling.
It is to be appreciated that while it is envisaged that the number of frequency bands and their bandwidth will usually be parameters that can be adjustable by the selection input, this is not always the case. More generally, the filterbank can be configured to enable one or more of usual parameters of a digital filterbank to be adjustable, and these can include: the number of bands; the width of each band; whether the bands have abutting band edges, overlap or are spaced apart; coefficients for both analysis and synthesis windows; whether there is any relationship between the analysis and synthesis windows; even or a odd stacking of bands; and the degree of oversampling above the critical sampling rate. Details of these parameters are set our below.
Preferably, the selection input enables at least one of the number of frequency bands and selection of stacking of said frequency bands in one of an even and an oddmanner to be selected, said number of frequency bands being equal to N, and the filter means comprises: (a) a first analysis filterbank means for separating said signal into the plurality of N separate frequency band signals; (b) processing means for receiving and processing each of said separate frequency band signals to provide N separate processed frequency band signals; and (c) a second synthesis filterbank means for receiving and recombining the N separate processed frequency band signals into a single output signal, wherein both of the first analysis filterbank means and the second synthesis filterbank means are connected to the selection input, the processing means being coupled between the first analysis filterbank means and the second synthesis filterbank means.
In another aspect of the present invention, the filterbank comprises a dedicated application specific integrated circuit (ASIC), said ASIC including the first analysis and the second synthesis filterbanks, and a programmable digital signal processor for controlling the number of frequency bands and the bandwidth of each frequency band, said digital signal processor being provided with the selection input.
The filterbank may be adapted to receive a single real monaural information signal, wherein said transform means generates non-negative frequency band signals and negative frequency band signals, said negative frequency band signals being derivable from the non-negative frequency band signals, and said processing means processes only said non-negative frequency band signals. Alternatively is adapted to filter an audio signal comprising first and second real monaural information signals which are combined into a complex stereo signal and wherein said transform means generates N combined frequency band signals, and wherein said processing means includes: (a) channel separation means for separating the N combined frequency band signals into the N frequency band signals corresponding to said first information signal and the N frequency band signals corresponding to said second information signal, each of said N frequency band signals comprising non-negative and negative frequency band signals; (b) first independent channel processing means connected to the channel separation means for receiving and processing each of said separate frequency band signals of said first information signal to provide a first set of N separate processed frequency band signals; (c) second independent channel processing means connected to channel separation means for receiving and processing each of said separate frequency band signals of said second information signal to provide a second set of N separate processed frequency band signals; and (d) channel combination means connected to the first and second independent channel processing means for combining said first set of N processed separate frequency band signals and said second set of N processed separate frequency band signals.
In accordance with another aspect of the present invention, there is provided a method of processing an information signal to selectively modify different frequency bands, the method comprising the steps of: (1) defining a filter frequency bandwidth to be analyzed; (2) dividing the filter frequency bandwidth into a plurality of uniformly spaced bands, said frequency bands being stacked in an even or odd manner and said frequency bands abutting, overlapping, or being spaced apart from one another; (3) filtering the information signal to separate the signal into a plurality of frequency band signals, each representing one of said uniform filter bands; (4) processing the frequency band signals; (5) recombining the signals of the individual bands to form an output signal; and (6) providing an input for enabling at least one of the following to be selected: (i) the number of frequency band signals, (ii) the bandwidth of said frequency bands, (iii) whether said frequency bands are stacked in an even or odd manner, (iv) whether said frequency bands abut, overlap, or are spaced apart from one another, and (v) a decimation factor by which said frequency band signals are downsampled.
In another aspect the method includes transforming the information signal into the frequency domain, providing N separate frequency band signals in the frequency domain, and effecting an inverse transform of the N separate processed frequency band signals into the output signal in the time domain.
Brennan Robert
Schneider Anthony Todd
Bereskin & Parr
dspfactory Ltd.
Kuntz Curtis
Ni Suhan
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