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E.2 Reconstruction of Contaminated and Masked Auditory Spectra

The objective of this section is to use time-frequency analysis methods to reconstruct contaminated or masked auditory spectra. In many acoustic applications, a spectral estimate of the source is found to be masked in a particular frequency band because of filtering in the environment, or because of jamming by competing sound sources. In these cases, it can be shown that certain techniques involving early auditory spectral representations, combined with time frequency analysis methods, may be able to recover the lost spectral information. One such approach can be formulated as follows, and is based on the assumption that the cochlear analysis of sound signal f can be thought of as the wavelet transform [.Yang Wang Shamma 1992.]

where t is time, is convolution, g is the impulse response for the cochlear filter G [.Allen 1985.], is a scale channel for some a>1, and . Including the effects of the nonlinear transduction and membrane leakage in the haircell, the cochlear output at each channel s is given by:

where is an instantaneous sigmoidal nonlinearity applied to followed by a lowpass filter H with impulse response h, and designates composition. The sound signal can be fully reconstructed from the cochlear outputs in the case of a lateral inhibitory network [.Morishita Yajima 1972, Shamma 1985 spee2.], by means of a Gabor irregular sampling theory developed in [.Benedetto Heller 1990, Benedetto 1992.]. This irregular sampling solution for sound reconstruction was given in [.Benedetto Teolis 1993.] cf., the solution in terms of alternating projections given in [.Yang Wang Shamma 1992.].

To specify this formulation for the case of omitted spectral data, and because of the form of , we shall study operators L defined by , where the filter H corresponding to h is no longer necessarily low pass. Our goal is to reconstruct f from cochlear-output-data Lf in the case when some spectral information is omitted from this data.

The first step in this project is to develop nonlinear wavelet packet algorithms for operators and where are essentially quadrature mirror filters. When these algorithms are formulated we shall design filters and dilation operators so that the frequency domain is decomposed in prescribed ways for an appropriate level of the nonlinear wavelet packet pyramid. This means that our frequency decompositions must include intervals associated with poor signal reconstruction, as might occur in a masking problem.

Technically, the aforementioned algorithms and filter design must provide a method of signal reconstruction. We have accomplished this in special cases [.Benedetto Saliani 1994.]. A full solution will involve the inversion of nonlinear ``cochlear output'' operators compatible with linear wavelet packet theory. Formally, the computation of f from available data and at this first level of our nonlinear wavelet packet pyramid is

where and designate the Fourier transform and inverse Fourier transform, respectively. The validity of this inversion formula depends on natural constraints imposed on the nonlinearity . A priori, there are problems of computational complexity especially when the omitted frequency information for the error estimation problem is at a low level of the pyramid. However, the linear wavelet packet reconstructions are -order algorithms, and there are reasons to believe some of this efficiency can be transmitted to the nonlinear case.



next up previous
Next: About this document Up: E. Generalized Time-Frequency Previous: E.1 Advances in



Didier A. Depireux
Mon May 19 16:21:14 EDT 1997