SPECTRUM_LMS

4元麦克风阵列去除语音混响，适合正在学习阵列去混响的同学们参考一下(Microphone Array Dereverberation)

多信号分离MUSIC的经典学习程序 对初学者帮助很大(it is a baisc and useful code of the multiple signal classfication for the beginners.)

语音合成软件，输入文字，自动生成声音文件，全部是源码(Speech synthesis software, input text, automatic generation of sound files, all the source code)

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.)

Chapter 2 Low Bit-Rate Speech Coding In this chapter an overview is given of speech coding techniques at several bit rates. Most of them use Linear Prediction. This overview is not meant to be complete its purpose is to make the reader somewhat familiar with Linear Predictive Coding which is necessary for a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and coding. In section 2.2 a description of speech production and speech sounds is given. Coders based on linear prediction can be considered as being based on a simple speech production model. This model is explained in section 2.3. Section 2.4 describes various speech coding algorithms and techniques. Section 2.5 briefly describes some measures for the quality of coded speech.(In this chapter an overview is given of speech coding techniques at several bit rates. Most of them use Linear Prediction. This overview is not meant to be complete its purpose is to make the reader somewhat familiar with Linear Predictive Coding which is necessary for a proper understanding of later chapters. Section 2.1 treats the subject of quantisation and coding. In section 2.2 a description of speech production and speech sounds is given. Coders based on linear prediction can be considered as being based on a simple speech production model. This model is explained in section 2.3. Section 2.4 describes various speech coding algorithms and techniques. Section 2.5 briefly describes some measures for the quality of coded speech.)

利用自相关或平均幅度差提取基音原理，帧长160，帧移80。求能量E、过零率Z和基音周期pitch；采用自相关法或平均幅度差法求取基音周期，基音周期范围（80~400hz）；画出能量、过零率和基音曲线。(Pitch principle is extracted by means of autocorrelation or mean amplitude difference)

基于hmm的识别程序 识别数字0-9，识别率全对(speech recognition based on hmm number 0-9 all right)

用自相关函数提取语音信号基频，提取音频文件的基频等高线(Use the autocorrelation function on segments of the signal (windowsize: 100ms) and compute the fundamental frequency. Use a max_time_lag of 100ms in the autocorrelation function and a window shift of 25ms. Create a fundamental frequency vector and plot the pitch contour.)

源－目标说话人声音转换的学位论文：源－目标说话人声音转换则是指将源说话人的语音模式转换成某个指定的特定目标说话人的语音模式，即保持源说话人原有的语音信息内容不变，使转换后的语音具有目标说话人的声音特点。(Source- target speaker voice conversion refers to the source speaker voice patterns into a designated target speaker-specific speech patterns, that is, the source speaker to keep the original voice message unchanged, so that the converted speech with the goal of the speaker s voice characteristics.)

做毕设是用到的gmm的matlab程序，测试过的 ，绝对可以用(Complete set is used to do the matlab program gmm, tested, can surely be)