搜索资源列表
qtphone-0.0.16
- QtPhone is a cross-platform telephone emulation application designed to be used with a PC, speakers, microphone, V.92 voice modem and a standard phone line. ISDN, xDSL and VoIP will be supported. Features include Caller Identification and Voice Mail.
后处理演示
- 视频音频处理演示文件，其中视频编解码包括XVID，H263P和H264，音频采用ILBC，音频做了AEC处理。-demonstration video audio processing documents, which include video codec XVID, H263P and H264, used ILBC audio, audio processing done by the AEC.
VideoNet
- 该程序可以用于两个人在LAN/Intranet(或者Internet)上进行视频会议。现在有许多视频会议程序，每个都有各自的性能提升技术。主要的问题是视频会议视频帧的尺寸对于传输来说太大。因此，性能依赖于对帧的编解码。 -the two procedures can be used in LAN / Intranet (or the Internet) for video conferencing. Now there are many video conference proceedings,
RadiusAAA
- radius AAA 源代码,基于radius协议的鉴权，认证组件，广泛用于电信行业和银行业。-radius AAA source code, based on the radius of the agreement authentication, authentication components widely used in the telecommunications industry and banking.
Artech House - Voice.over.802.11
- 介绍通过无线环境来传输语音的一本英文原版书籍，这里用到得无线协议是802.11 -introduced through wireless environment to transmit voice of an English original books, here used in the 802.11 wireless protocol is
opengk_20020522_win32
- 运行于WINDOWS下的open gk 源码，可以用于视频会议系统的GK-running on Windows under the open-source gk, can be used for video conferencing system GK
zlib123
- The deflation algorithm used by gzip (also zip and zlib) is a variation of LZ77 (Lempel-Ziv 1977, see reference below). It finds duplicated strings in the input data. The second occurrence of a string is replaced by a pointer to the previous
IP电话实现程序
- ip电话范例,采用sip协议簇,支持rtp, g.729系列编解码标准,rtp采用md5加密-ip phone model used sip agreement cluster support rtp, codec G.729 series standards, rtp encryption using md5
H263code
- 应用于视频会议系统的视频编/解码技术,可以在5种标准图像格式sub-QCIF\\QCIF\\CIF\\4CIF\\16CIF上工作-video conferencing systems used video encoding / decoding technology, the five standard image formats sub-QCIF \\ QCIF \\ CIF \\ are applicable \\ working on 16CIF
Ip-Ipyuying
- 一个IP-IP的语音电话源代码,应用了第三方控件,经测试可用,供大家参考-an IP-IP voice source, a third-party control applications, the test can be used for reference
ipdephi
- ip电话源代码，可以免费打电话的，仅供研究之用 -ip phone source code, free phone, used for research purposes only
LATransmitter
- 用JMF 实现语音信息发送端程序，从麦克风中抓取语音信号，使用sessionManager进行会话管理和传输 -with JMF voice messaging software, which crawls from the microphone voice signal sessionManager used for session management and transfer
jvoiplib-1.4.1
- 这是一个外国人写的语音聊天代码,其中使用了RTP协议-This is a foreigner's voice chat write code, which used the RTP
libcu30-1.0-770.src
- libcu 3.0 codec 源代码，用了卡奈尔大学的算法-libcu 3.0 codec source code used by the University of algorithm 207,000
AACPLUS
- 语音编码，极好的视听编解码标准，可以用在iptv等领域。-voice coding, excellent audio-visual codec standards, can be used in areas such as IPTV.
H323协议包产生工具(WIN)
- 需要的可以用到 ！专业性强！-needs can be used! Professional strong!
speechVAD
- 活动语音检测，语音开发中的重要算法，尤其是在voip中广泛使用。该算法简单明了。-voice activity detection, voice development of the algorithm, especially in the widely used voip. The algorithm is simple and clear.
IPphone2.0
- 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
p2pshipin
- 点对点视频会议程序:VideoNet 该程序可以用于两个人在LAN/Intranet(或者Internet)上进行视频会议。现在有许多视频会议程序，每个都有各自的性能提升技术。主要的问题是视频会议视频帧的尺寸对于传输来说太大。因此，性能依赖于对帧的编解码-point-to-point video conferencing : VideoNet the two procedures can be used in LAN / Intranet (or Intern et) on video co
TalkG726
- 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