xmms This is a 3D audio plugin. Algorithm is simple, highly efficient, if used t...

关于dtw的书籍，电子书html格式，dtw 用于语音识别 -on the books, e-books html format, dtw for Speech Recognition

语音识别专辑三- Speech recognition special edition three

一个HMM的Matlab实现方法，可实现孤立词语音识别-The Matlab realize a HMM method can realize an isolated word speech recognition

语音识别（Speech Recognition）是让机器通过识别和理解过程把语音信号转变为相应的文本或命令的技术。在课题中，通过采用DTW（Dynamic time warping, 动态时间伸缩）算法，对实现孤立词的识别进行了初步探讨和研究，实现了在MATLAB软件环境下孤立词语的语音识别，并针对DTW的主要特点及不足做出了总结。-Speech Recognition (Speech Recognition) machines is through recognition and understanding of a process which put the voice signal into the corresponding text or command technology. The issue, through the use DTW (Dynamic time warping, dynamic time telescopic) algorithm to achieve an isolated word recognition conducted a preliminary exploration and study, the realization of the MATLAB software environment isolated words of Speech Recognition and DTW against the main features and deficiencies make summarized.

一款安卓版的语音小助手，调用了讯飞的API，实现了用语音指令打电话、发短信、查看日历、搜索、打开应用等数十项功能，类似于APPLE的SIRI，用Android studio开发。

不错的FIR滤波器源代码！-good FIR filter source code!

本书给出80个语音处理习题和4个语音处理程序，对相关人员有很大价值-the book is 80 speech processing and Exercise 4 speech processing procedures, the relevant personnel be of great value

采用小波变换实现带噪语音信号的增强，供参考-Wavelet transform realize Noisy speech signal enhancement, for reference

系统辨识与自适应控制MATLAB仿真CH5,好东西大家分享-System Identification and Adaptive Control of MATLAB simulation of CH5, a good thing to share with you

Levison-Durbin 语音信号处理中的线性预测编码LPC 理论、格型滤波器以及求解现行预 测方程的算法，可以实现对语音信号重要元素的分析、合成甚至识别。 基于现有的实验平台，我们可以利用 Matlab 函数来获得几个固定语音元素（如元音） 的模型系数，LPC 得到的系数组成 IIR 滤波器。利用冲击脉冲 序列作为输入，我们就可以得到原来的语音。这是一种简单的语音合成功能。-Levison-Durbin speech signal processing in linear predictive coding LPC theory, lattice filters, as well as the current prediction equation solving algorithm, can achieve an important element of the speech signal analysis, synthesis or recognition. Based on the existing experimental platform, we can use Matlab function to obtain the number of fixed-voice elements (such as vowels) model coefficients, LPC coefficients are the composition of IIR filters. Shock pulse sequence used as input, we can get the original voice. This is a simple voice synthesis.