FDM array-based dual channel speech enhancement method;

A study of an improved minimum mean-square error speech enhancement algorithm;

Modified speech enhancement algorithm based on least mean square error estimation;

An improved method for signal enhancement based on ICA;

The extracted fault characteristics of transient zero sequence current is looked as condition attributes and the multiple of signal enhancement is.

Methods of waveguide-multilayer(WM) optical disk design and tracking signal enhancement are presented.

Method of solving LPC coefficients of speech signals;

the Character Analysis of Speech Signal with Time and Frequency Based on MATLAB;

Method to determine delay time of speech signal in reconstructed phase space;

Blind source separation of speech signals based on the amplitude correlation of neighbour bins;

The Study of Blind Source Separation of Speech Signals;

Based on the analysis of average displacement (AD) method for selection of time delay in phase space reconstruction of speech signals, this paper modifies the original measurement of AD method to get a better criterion for proper time delay selection, which has a sound theoretic basis and can produce better effect.

The design of the voice signal real time noise cancellation software;

This paper elaborates on the basic principle of linear preduction inanalysing voice signal technology and solves the effective calculation method about thecoupled equations of linear preduction by using self-interrelative theory.

A DSP control based voice systems, with TMS320VC5402bas a microcontroller, realizes the functions of voice signal acquisition compression and data transmission.