## IEEE Digital Signal Processing Workshop |

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### Contents

Separation of CoChannel FMPM Signals Using the Discrete PolynomialPhase Transform | 3 |

Extraction of the Inphase and Quadrature Components from Oversampled Bandpass Signals | 11 |

Filter Design to Guarantee Convergence of the Pseudolinear Regression IIR Adaptive Algorithm | 19 |

Copyright | |

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### Common terms and phrases

Acoustics adaptive filter amplitude analysis filters applications approach approximation beamformers behavior channel coefficients complex components computational constraints convergence corresponding covariance matrix decomposition defined delay denotes detector digital filters Digital Signal Processing domain eigenfilter eigenvalues eigenvector Engineering equalizer equations error estimate Figure filter banks filter design finite FIR filter Fourier transform frequency response Gaussian given IEEE IEEE Trans implementation impulse response initial phase input signal inverse iteration lattice least squares limit cycle linear LMS algorithm Lyapunov exponents M-channel Matlab method modulated multichannel n-width noise nonlinear obtained optimal orthogonal output paper parameters performance polyphase problem Proc properties pseudoinverse quantizer recursive Remez algorithm rithm rotation samples scale sequence shown Signal Processing simulation sinusoidal solution spectral speech stopband structure subband subspace synthesis filters technique time-frequency time-varying tion transfer function transition band update values vector wavelet weighting zero