G.723语音编码 The source code has been modified by simplifying the ADPCM algorithm...

这一个让你的网页拥有视频和语音功能的一个ActiveX控件!用起来很简单-this one allow you to the website has video and voice capabilities of an ActiveX control! Together with a very simple

TI arm芯片在ivr ewarm中的flash下载驱动源码，和生成文件，直接可用，比较少见。-TI arm chip ivr ewarm in the flash drive to download source code, and generate documents directly available, relatively rare.

最后打包版本小错误，来源新的实验包，音箱命名的任务，正在进行的会议系统，javaCV（OpenCV在JAVA）的测试，电话diarization（ANR项目MEDIA）。LIUM_SpkDiarization-4.2.jar，包含一个完整的jar文件，准备使用的编译版本，包括所有的类（LIUM_SpkDiarization和外部类）和GMM资源（Java的1.6）另外，下载LIUM_SpkDiarization-4.2.src.tgz gzip压缩的tar

求语谱图的matlab程序，非常有用-Spectra for the matlab language program, very useful

语音识别与合成系统. 里面有详细的介绍与程序-Speech Recognition and Synthesis System. There are a detailed introduction and procedures

赤木正人教授在论文“Development of an F0 control model based on F0 dynamic characteristics for singing-voice synthesis”中提出的f0 vibrato模型的matlab实现-Professor Masato Akagi paper "Development of an F0 control model based on F0 dynamic characteristics for singing-voice synthesis" presented f0 vibrato model matlab implementation

语音识别，跳用JAVA里写的类，a simple test project-voice recognition, dancing with Java, written category, a simple test project

基于分形的纹理图像的金字塔处理方法,搞分形的可以-A grain image synthesis method depend on Pyramid algorithmic. It is useful for guys doing separate shape

为了实现高速语音特征参数的提取，在分析了美尔频率倒谱特征参数提取算法的基础上，提出了算法的硬件 设计方案，介绍了各模块的设计原理。该方案增加了语音激活检测功能，可对语音信号中的噪音帧进行检测，提高了特征参 数的可靠性。-In order to achieve high-speed voice characteristic parameter extraction, in the analysis of Mel frequency cepstral feature extraction algorithm is proposed based on the algorithm the hardware Design, describes the design principles of each module. The program increased the Voice Activity Detection feature, which allows voice signals to detect the noise frame to improve the characteristic parameters The number of reliability.

语音信号的频域处理,语音虽然是一个时变、非平稳的随机过程。但在短时间内可近似看作是平稳的。因此如果能从带噪语音的短时谱中估计出“纯净”语音的短时谱，即可达到语音增强的目的。由于噪声也是随机过程，因此这种估计只能建立在统计模型基础上。利用人耳感知对语音频谱分量的相位不敏感的特性，这类语音增强算法主要针对短时谱的幅度估计。 -voice signals in the frequency domain processing, voice is a time-varying, nonstationary random process. But in a short period of time can be approximated as smooth. So if Noisy Speech from the short-term spectrum estimate "pure" voice of the short-term spectrum, and reached speech enhancement purposes. As the noise is random process, which can only be estimated based on statistical models based on. Use ear perception of voice spectrum component of the phase sensitive to the characteristics of such speech enhancement algorithms targeted at the rate of short-term spectral estimation.