FSK demodulation telling how to do it after AWGN noise.

在做和语音信号处理相关的项目时要用到的语音工具箱,比如语音识别,说话人识别等,里面有一些已经封装好的小函数,可供大家编程序的时候直接调用-In speech signal processing to do and projects related to voice use toolbox, such as speech recognition, speaker recognition, etc., there are some good small function package for everyone when programmed direct call

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))

vorbis开发包及源码-vorbis development kits and source

This code is used to remove silence from a voice signal so that the recognition in voice recognition will give good result. The method is Scale on threshold applied to the envelope for detecting scilence periods. The actual threshold is computed by multiplying threshin" by the median value of the envelope samples. The default is (.25), whichis 25 of the median envelope value over input signal .-This code is used to remove silence from a voice signal so that the recognition in voice recognition will give good result. The method is Scale on threshold applied to the envelope for detecting scilence periods. The actual threshold is computed by multiplying threshin" by the median value of the envelope samples. The default is (.25), whichis 25 of the median envelope value over input signal .

一种噪声环境下连续语音识别的快速端点检测算法，格式ｖｉｐ-a noisy environment for the rapid identification voice endpoint detection algorithm format vip

This PDF gives the IVR router configuration details

非常不错的，文字转拼音的小程序，代码风格很好。-Very good, Text-to-phonetic small procedures, a good code style.

用DSP进行语音压缩的经典开发实例（PCI总线)-A typical sample of developing which use DSP to compress speech signal

英国爱丁堡大学的语音处理系统，可以找到大部分语音处理/识别/合成所需要的算法-University of Edinburgh voice processing system that can find most of voice processing/recognition/synthesis algorithms needed

本文件包含几个进行语音端点检测的小程序。运行时直接复制到matlab的m文件即可。经本人调试正确无误。-This document contains some of the voice activity detection applet. Directly copied to the run-time of the m file to matlab. After I debug correct.