EVC platform recording procedures. Mack good configuration can be tape recorded....

语音信号处理方面的源代码，简单的语音合成软件（text to speech）-voice signal processing of the source code, a simple speech synthesis software (text to speech)

This PDF gives the IVR router configuration details

语音识别芯片LD3320的口令模式，程序代码，仅供参考

一个简易的端点检测程序，已经实现，包括基于过零率、短时能量的双门限判决方法。-a simple endpoint detection procedures has been achieved, including on the basis of zero rate Short-term dual energy threshold judgment method.

LPC 10 Code,比较经过lpc编码与没有经过lpc编码的语音差别，用自相关法计算lpc参数。-LPC 10 Code, comparison, after encoding and not LPC LPC voice coding differences, using auto-correlation method to calculate LPC parameters.

g.711 C代码-G.711 C code

TTS应用，完成朗读功能，你可以在源码基础上用半个小时写出同样的程序。-TTS application, complete the reading function, you can source the basis of half an hour to write using the same procedure.

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

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.

数字音频格式有很多种，其质量与采样频率和采样精度两个参数有关。频率的表示单位为赫兹〔Hz〕，它表示每秒采样次数。采样频率越高，音质就越好。采样精度为每次采样所存储的数据数量，它决定每个数字信号所能够表示的离散振幅的数量。存储每个样本的数据越多，音质就越好。但是高品质的声音需要占用大量的内存和磁盘空间。考虑到网络带宽，在Internet连接上传输就需要花费很长的时间。对于Applet来说，保证声音文件的最小化是极为重要的。-digital audio format there are many, the quality and frequency of sampling and sampling accuracy of the two parameters. The frequency of Hertz unit of [Hz], it said the number of samples per second. The higher sampling frequency, the sound quality better. For each sampling precision sampling by the amount of data storage, it was decided that each digital signal can be expressed as the number of discrete amplitude. Each sample storage of more data, better sound quality. But high-quality voice would take a lot of memory and disk space. Consideration of the network bandwidth, the transmission on the Internet connection on the need to spend a very long time. Applet for instance, to ensure that voices of the smallest document is very important.