ETSI发布的用于研究分布式语音识别的前端降噪算法

使用DirectSound和Microsoft agent实现的语音复读机- Uses DirectSound and Microsoft agent realization pronunciation duplicate reading machine

语音识别与合成入门5，word或压缩包中内有matlab源码-speech recognition and synthesis 5 entry, word or compressed packages within Matlab FOSS

VOIP行业的H323协议终端软电话的源代码-VOIP industry H323 protocol source code for the terminal soft-phone

语音端点检测，是语音压缩时提取特征参数的一个必要的过程，这个程序主要是对这个过程的一个小的程序。希望对学语音的有帮助-Extreme point detecting is a primily process to get the eigenparameter when you compressing the speech.This program is a small one,it will help for the learner.

具有浮点指令的g.729语音压缩编码，符合ITU-T G.729 Annex C 规范，通过修改makefile可以支持windows/linux/solaris等操作系统，仅需要很少的更改就可以应用到voip领域-With floating-point instructions G.729 voice compression coding, in line with the ITU-T G.729 Annex C specification by modifying the makefile to support windows/linux/solaris and other operating systems, only a small change on the field can be applied to voip

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.

用于语音识别的离散隐马尔可夫模型程序（非特定人，多个观测序列），已调试通过。-used discrete speech recognition Hidden Markov Model (non-specific, a number of observation sequence), has been through testing.

CCS下调试通过的基于5416DSP及AIC23的语音数据调出、并播放。-CCS under the debugger through the AIC23-based 5416DSP and out of voice data, and play.

EC的语音编解码-EC voice codec

利用全极点模型产生语音信号.rar-use all-pole model generated voice signal. Rar