IEEE Digital Signal Processing WorkshopIEEE, 1994 - Signal processing |
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 |
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Acoust adaptive filter amplitude analysis applications approach approximation attractor beamformers behavior channel coefficients complex components Computer constraints convergence corresponding defined delay denotes detector digital filters Digital Signal Processing domain eigenvalues Engineering equalizer equations error estimate example Figure filter banks filter design finite FIR filter Fourier transform frequency response given IEEE IEEE Trans implementation impulse response input signal inverse iteration limit cycle linear LMS algorithm Lyapunov exponents M-channel magnitude Matlab matrix MaxiCode method modulated n-width nonlinear obtained optimal orthogonal output paper parameters performance phase polynomial problem Proc processors properties quantizer reconstruction recursive reference set Remez Remez algorithm samples scale sequence shown Signal Processing simulation sinusoidal solution spectral speech squared stopband subband subspace synthesis filters techniques time-frequency time-varying tion transfer function transition band trispectrum update values vector wavelet zero