Delphi PSK无线通信，基于COM的原代码的声卡…

LMS算法实现自适应滤波 clear close all clc N=10000 设置仿真长度 信号产生参数设定 a1=-0.195 a1=-1.5955 a2=0.95 R0=[1,a1,a2 a1,1+a2,0 a2,a1,1] p=[1,0,0] r=inv(R0)*p 计算理论自相关函数 R=[r(1),r(2) r(2),r(1)] 生成理论自相关矩阵 p1=[r(2),r(3)] 生成互相关 h=inv(R)*p1 计算维纳解 Jmin=r(1)-h *p1 计算维纳解时最小均方误差 u=1/sum(eigs(R- LMS算法实现自适应滤波 clear close all clc N=10000 设置仿真长度 信号产生参数设定 a1=-0.195 a1=-1.5955 a2=0.95 R0=[1,a1,a2 a1,1+a2,0 a2,a1,1] p=[1,0,0] r=inv(R0)*p 计算理论自相关函数 R=[r(1),r(2) r(2),r(1)] 生成理论自相关矩阵 p1=[r(2),r(3)] 生成互相关 h=inv(R)*p1 计算维纳解 Jmin=r(1)-h*p1 计算维纳解时最小均方误差 u=1/sum(eigs(R))

本软件需要东进电话卡的支持 代码完全可靠-East into phone cards are supported entirely reliable code

语音信号端点检测的实验研究论文,希望对大家有点帮助-voice signal endpoint detection of experimental studies, we hope to help a bit

matlab的时频分析工具箱，希望在对信号的分析上可以帮助到大家。-the time-frequency analysis matlab toolbox, hope that the analysis of the signal can help to everyone.

利用杜宾算法求lpcc系数，非常有用-Durbin algorithm for the use of lpcc coefficient, very useful

线性量化。读取WAV文件，用不同的比特数量化，抽样频率在8khz-Linear quantization. Read WAV files, with different number of bits, the sampling frequency of 8kHz

这是一个从音频信号里提取特征参量的程序，从事此方面的同志，不妨参考-This a signal from the audio extract features parametric procedures, engaged in this regard, the comrades might reference! !

60秒多段语音录放芯片APR9600.htm-60 seconds of voice recording and multi-chip APR9600.htm

语音合成软件，包括Freephone两个模块和Mbrora

本文将现实中采集到的一段语音加上了高斯白噪声，利用了LMS算法，将噪声滤除，仿真效果良好-This article will be collected in reality a section of voice coupled with a Gaussian white noise, using the LMS algorithm, the noise filtering, simulation results