使用directshow来进行mpeg

用C＋＋语言写的开源SIP协议栈源码 编译环境为vs.net 2003-C++ language to write open-source SIP protocol stack source compiler environment vs.net 2003

MRCP协议V1和V2版实现源代码，这是做TTS和ASR的朋友的好东东。-MRCP protocol version V1 and V2 realize the source code, which is to do TTS and ASR Dongdong good friends.

OSIP最新代码-CBS.MW latest code

JRTP可用于组播也可用于单播，JRTP主页上有一个关于组播的客户端和服务器的例子，还有一个单播的客户端和服务器在同一个程序的例子，本人一直想做一个单播的客户端和服务器分开的例子，在想了一两天后终于想通，做完后发上来，秉承本人一贯风格，注释详尽、附上测试结果，倒不是什么难的东西，只是牛人都不愿做这样的东西，但还是对初学者非常有用，希望得此帮助的给个好评，只此一家，别无分店。-JRTP can be used for multicast can also be used for unicast, JRTP have a home page on the multicast client and server example, there is a unicast client and server program in the same example, I have always wanted to do a single broadcast of the separate client and server example, thinking about it for a finally realized after 12 days, after finishing up, and to inherit has been my style, notes in detail, together with test results, it is not a difficult thing, but cattle were are unwilling to do this kind of thing, but still very useful for beginners, I hope to have the help of praise, only this one, there is no branch.

VoIP正在arapidly不断发展的技术，可以彻底改变可能的电信业。当wirelessdata网络实施，网络电话可能被证明是有助于收敛ofexisting固定和移动电话网络与快速增长的有线andwireless数据网络。服务的能力andQuality是两个需要无线VoIP的商业部署之前beresolved最重要的问题。本文开头的数据adetailed解释网络与问候网络建模与流量分布在一起。的VoIP，WLAN和VoIP在WLAN已经解释的优势，劣势和实施基于WLAN的VoIP的问题。不同的VoIP传输系统：RTP，RTSP，SIP和H323也进行了审查。一个8节点的移动电话系统hasbeen使用NS2和实验设计已经进行测量thesystem在提供服务的质量方面通过一系列的因素，包括多个节点，包大小和客流率。如延迟，吞吐量和packetlost参数已经测量通过WLAN来评估的VoIP的服务质量。基于一个NS2代码和differentparameters如延迟，吞吐量和数据包丢失的打印输出包括以supportthis论文。

jpeg视频编解码程序，包括dct变换、量化、vlc等过程，针对视频压缩-jpeg video codec procedures, including Extra transform, quantization, VLC and so on, for video compression

SIP Stacks dissipate2: C++ Linux, requries the qt-library, License: GPL Homepage: http://www.wirlab.net/kphone/ A enhanced dissipate, is part of the kphone distribution. -SIP Stacksdissipate2: C++ Linux, requries the qt-library, License: GPL Homepage: http://www.wirlab.net/kphone/ A enhanced dissipate, is part of the kphone distribution.

G.722语音编码实例化解析VC语言描述!-G.722 speech coding examples of analytic description language VC!

voip技术现在已成为国内外的热点话题，实时语音通信对于我们来说是很重要的。voip技术的应用当然也不在话下。 -voip technology has become a hot topic at home and abroad, real-time voice communication for us is very important. Voip technology applications course presentations.

2.0说明 1.0支持系统的录音多种格式,有一定的延时,没有过滤杂音功能. 2.0已取消支持系统的录音多种格式,使用PCM采集数据G711A压缩格式(8000HZ单声道16位格式录音每秒以8K完成数据,16000HZ单声道16位格式录音每秒以11K完成数据,音质相当好),延时降低到最小100-500MS以内,不会随时间增加而增加延时(如果是说话测试一直保持200MS的延时,如果是用播放歌曲来测试,有自动校正延时功能,恢复成200MS的延时,恢复过程中不会中断歌曲的播放,只是小小加快唱歌的速度,听觉不会觉察出来),加入了过滤杂音功能. 2.0安装 将ACMG711.dll复制到系统文件夹里 再安装控件 -2.0 1.0 Note recording system to support multiple formats, a certain delay, no noise filtering function. 2.0 support system has been abolished recording formats, the use of PCM data acquisition G711A compression format (8000HZ 16 mono recording format to 8K per second complete data 16000HZ mono recording format 16 to 11K per second complete data quality is very good), Delay reduced to the minimum 100-500MS within not with time delay increases (if word has been tested 200MS delay, If it is used to test play the songs, the automatic correction function of delay, 200MS resume into the delay, the recovery process will not be interrupted play songs, singing just a small speed up