duoxinhaofenli

有关语音的matlab源程序,其中包含语音的频谱分析等.(Matlab source code on the voice, including voice spectrum analysis.)

通常语音信号在增强时会出现混响现象，演讲者为了消除背景混响，不得不频繁地偏转头部的方向，这样会造成脉冲响应的不断改变。我们结合盲解卷法和频谱消去法来提高逆滤波器的滤波效果。我们利用输入语音信号间的相关系数矩阵计算出稳定、精确的室内脉冲响应的逆滤波器，而这些输入信号无需测量室内的脉冲响应就能被观测到。逆滤波能够消除早期的反射，这些反射包含混响中的绝大部分能量。之后，用频谱消去法来抑制逆滤波后的信号的尾部混响。本方法在实际适应性方面的表现通过具体的实验进行了验证，结果表明盲解卷法和频谱消去法的结合相较于单独使用一种方法，能够提供一个更优越的演说环境。(Usually voice signal will be enhanced when the reverberation phenomenon, the speaker in order to eliminate the background reverberation, and had to frequently deflect the direction of the head, this will cause the impulse response of changing. We combine the blind deconvolution and spectral elimination method to improve the filtering effect of the inverse filter. We use the correlation coefficient matrix between the input speech signal to calculate the stable, accurate inverse filter of the indoor impulse response, and these input signals do not need to measure the room impulse response can be observed. Inverse filtering to eliminate early reflections, these reflections contains most of the energy in the reverberation. After the spectrum elimination method to suppress the tail reverberation of the signal after the inverse filtering. Performance in the actual adaptation of the method by specific experimental validation results show that the blind deconvolution and spectrum to eliminate )

matlab在语音信号处理中的使用，包括基频提取，语音识别，语音增强等(The application of MATLAB in speech signal processing including pitch extraction, speech recognition, speech enhancement and so on.It uses matlab.)

LMS多麦克风语音降噪的主程序是lmsspdn.m(Multi-microphone noise reduction LMS voice is the main program lmsspdn.m)

这是一个与语音识别与端点检测有关的文件,是用C++写的源程序,相信对大家有用(This is a speech recognition with the Endpoint Detection of the documents are written in C source files, it may be useful to think)

适用于对各种语音信号的分帧，将语音信号分段处理(Applicable in a variety of speech signal sub-frame, the speech signal Break Processing)

语音库开发工具源码，例程程序创建机读SAPI.SpVoice对象开发语音库。(Speech development tools source , routine program to create machine-readable SAPI.SpVoice speech object development .)

语音增强电子书，国外经典教材，自适应滤波，卡尔曼滤波，最小均方算法等等(speech enhancement ebook)

基于语音的人类年龄自动估计算法实现，效率比较高(voice-based human age estimation algorithm automatically to achieve more efficient)

对语音信号基频的提取对于语音信号的分析有一定作用(Basic frequency of speech Analysis )