web视频聊天室

RealNetworks公司的HELIX的audio.部分-RealNetworks, the company verified our audio. Some

第四章代码，Mpeg4的编码与解码，G.721,G.723,G.728,G.729的代码实现-Chapter IV code, Mpeg4 encoding and decoding, G.721, G.723, G.728, G.729 code to achieve

视频会议系统,进行网络视频会议的源代码，应该很详细了，数据源可能得进行改动一下-video conferencing systems, video conferencing network for the source code should be very detailed, data source may need to engage you change

在Internet上传播声音的源程序- On Internet carry over sound source program

Linux下开发的一款网络电话-Linux built a network phone

一个ip电话实例的实现，以vc为开发环境。-an example of the realization of telephone, vc for the development environment.

使用Rtc/RTCP技术来传输VIDEO和SOUND的程序(需要到MICROSOFT下载最新的RTC CLIENT SDK API才能编译，VC7不需要下载)-use Rtc/RTCP technology to transmit VIDEO SOUND and procedures (need to download the latest Microsoft RTC Client API SDK can be built, VC7 do not need to download)

G726局域网语音通话程序和源代码 这是使用G726语音压缩（16kbps）和RTP进行传输的程序，因为我没有带WIFI的PPC，所以每个程序都是单独测试的，PC端和PPC端分别都工作正常。 G726编解码算法来自OpenH323.传输使用的RTP可以在RTP程序中找到讲解，这个程序主要是G726的函数。将整个 G726封装为g726_Encode和g726_Decode两个函数，参数为压缩和解压数据存储的地址指针，可以将960字节压缩到120字节和将 120字节解压为960字节。这里G726使用的时候，音频设置为8kHz，16位量化，单声道。 使用方法很简单，只用两端各自输入对方的IP，然后按下“开始对话”，就可以进行语音通信了。 PPC端的运行比较稳定，已经进行了自收自发近一个小时连续工作的测试，非常稳定，话音清晰。-G726 LAN voice calls procedures and the source code is the use of voice compression G726 (16kbps ) and RTP for the transmission process, because I did not bring WIFI the PPC, there are separate procedures for each test, PC- and PPC-were all normal work. G726 codec algorithm from the OpenH323. RTP transmission can be used in the process to find RTP stresses Xie, this procedure is mainly a function of G726. Packaging whole G726 for g726_Encode and g726_Decode two functions, Parameters for the compression and decompression of data storage address pointer, 960 bytes can be compressed to 120 to 120 bytes and bytes to

openh323 通用库 -Universal library for openh323

Interfacing the TMS320C54xx to the TLC320AD535 and TLC320AD545 Codecs