语音识别方面的资料，大部分是引文资料非常有用

语音识别中经典的HMM算法，包括产生序列、测试和参数训练，由C语言编写。- In speech recognition classical HMM algorithm, including has the sequence, the test and the parameter training, compiles by the C language.

用Ｅｃｌｉｐｓｅ开发的ＭＩＳ系统。这个系统采用Ｈｉｂｅｒｎａｔｅ作为数据持久层，采用Ｓｔｒｕｔｓ模式开发，希望对中间件感兴趣的朋友跟我讨论-use Eclipse development of the MIS system. The system uses Hibernate as a data persistence layer, using Struts model development, Middleware hope to interested friends talked to me

在linux平台下,纯c写的盲信号分离的代码.它采用基于卷积混合的盲信号分离算法,不但可以分离人工合成的混合信号,而且对于真实环境中的卷积混合的语音信号也能够分离.在本程序中,包含了两个测试文件,makefile后便可以直接使用. 另外值得一提的是,压缩包里包含有一些语音处理方面的常用函数.例如fft变换,读取写入wav文件,以及一些常用的一维向量和二维矩阵变换的函数.这些可以直接应用在其他应用程序里去.-in linux platform, net write c Blind Signal Separation code. it is based on convolution of mixed Blind Signal Separation algorithm, not only can be separated from the artificial synthesis of mixed-signal, but for the real environment convolution of mixed voice signals can be separated. in this procedure, the tests included two documents, makefile can be used directly after. Also worth mentioning is that Compression bag containing some of voice processing functions commonly used. fft transform, for example, to read write wav file, and some of the commonly used one-dimensional and two-dimensional vector matrix transformation functions. these can be directly applied to other applications going.

《FUNDAMENTALS OF SPEECH RECOGNITION（语音识别基本原理）》（英文），大家可以好好学习下-"FUNDAMENTALS OF SPEECH RECOGNITION (voice consensus other basic tenets) "(English), you can learn where

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.

实现了精简的FFT语音压缩 采取了一种新的算法有一定的参考价值-Implementation of speech compress used by simplified FFT algorithm.It is a new one,and useful to you.

根据midi音乐的相关参数生成midi音乐文件-midi music under the relevant parameters generated midi music files

本软件需要东进电话卡的支持 代码完全可靠-East into phone cards are supported entirely reliable code

G.723语音编码 The source code has been modified by simplifying the ADPCM algorithm to increase running speed. Here, the quantizer scale factor, y(k), is determined by "unlocked" factor yu(k) only. The factor yl(k) and speed-contral factor al(k) are ignored.-G.723 speech coding The source code has been modified by simplifying the ADPCM algorithm to increase running speed. Here, the Quantizer scale factor, y (k), is determined by "unlocked" factor yu (k) only. The factor yl (k) and speed-contral factor al (k) are ignored.

老 外 写 的 虚 拟 鼓 原 代 码-foreigner wrote the original code virtual drum