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本程序介绍了一种改进的LMS算法-均一化的LMS算法，与传统的LMS算法比较收敛性更好，并由程序给出了比较图(This procedure introduced an improved LMS algorithm- homogenization of the LMS algorithm, with the traditional LMS algorithm convergence better, by the procedures given comparison)

自己用matlab编的语音合成识别程序，验证好使。(Yourself using matlab speech synthesis recognition program, verify so.)

短时自相关和短时平均幅度差计算基音周期的matlab程序，附带有原语音信号(Short-term autocorrelation and short-term average deviation of pitch period matlab program, comes with the original speech signal)

I:CSDN技术中心 进程管理器---一个SDK的例子(I : CSDN Technology Center process management-- for example an SDK)

本书系统地阐述了语音信号处理的原理、方法、技术和应用，同时给出了部分内容对应的MATLAB仿真源程序。全书共12章，第1章至第7章是基本理论部分，包括语音信号的数字模型、语音信号的短时时域分析和频域分析、语音信号的同态处理、语音信号线性预测分析和矢量量化；第8章至第12章是应用部分，包括语音编码、语音合成、语音识别、语音增强和语音处理的实时实现。本书内容全面，重点突出，原理阐述深入浅出，注重理论与实际应用的结合，可读性强。(This book describes the speech signal processing principles, methods, techniques and applications, and gives the corresponding part of the contents of the MATLAB simulation source. The book is 12 chapters, Chapter 1 to Chapter 7 is the basic theoretical part, including voice signal digital model, speech signal analysis in time domain and frequency domain analysis, speech signal homomorphic processing, speech signal analysis and linear prediction vector quantified Chapter 8 to Chapter 12 is the application of parts, including speech coding, speech synthesis, speech recognition, speech enhancement and voice processing, real-time implementation. The book is comprehensive, focused and Rationale layman, focusing on the combination of theory and practical application, readable.)

在matlab平台利用GUI界面实现录音，播放。并对语音信号进行频谱分析，特定人语音识别，混响，混频等基础功能。(On the MATLAB platform, the GUI interface is used to record and play. And the speech signal spectrum analysis, specific speech recognition, reverberation, mixing and other basic functions.)

说话人识别系统,界面友好，可以对实时录音的说话人进行识别(Speaker recognition system, user-friendly, real-time recording can be carried out to identify the speaker)

matlab编写的PCM脉冲编码调制simulink仿真模型，包含对不同频率语音信号的仿真。(Matlab prepared PCM pulse code modulation simulink simulation model, including the simulation of different frequencies of speech signals.)

语音识别 安卓源代码 和siri一样的音控软件 轻松掌握其中的原理(Speech recognition Anzhuo Yuan code and siri sound control software easily grasp the principle)

为均衡带限信号所引起失真的横向或格型自适应均衡器(其中横向FIR系统长M=11), 系统输入是取值为±1的随机序列 ，其均值为零；参考信号 ；信道具有脉冲响应： 式中w用来控制信道的幅度失真(w = 2～4，例如，取w = 2.9,3.1,3.3,3.5等),而且信道受到均值为零、方差为 (例如，取 ，相当于信噪比为30dB)的高斯白噪声 的干扰。试比较基于下列五种算法自适应均衡器在不同信道失真、不同噪声干扰下的收敛情况(对应于每一种情况，在同一坐标下画出其学习曲线)： 横向/格-梯型结构LMS算法[1-4] 横向/格-梯型结构RLS算法[1-4] ()