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Description: 基于UDP的网络音频系统的研究与实现,有很好的参考价值!-UDP-based network audio system with the study, and it is a good reference value!
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Size: 728064 |
Author: 张艳杰 |
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Description: 点对点视频、音频聊天的小工具。
该工具使用Windows的VFW库进行视频采集、压缩。
该工具中有一些自封装类库可提供大小学习之用；
-point-to-point video and audio chatting small tools. The use of tools for Windows VFW video capture, compression. The tool has some self Packaging Class size will provide learning;
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Size: 3326976 |
Author: onlyfish |
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Description: AgileIM 是一款类似QQ、MSN的聊天软件，这个软件却是我“无意插柳”的结果，AgileIM 的“前身”原本只是为了测试我的通信框架ESFramework的通信能力，没想到，做了两个星期后则个用于测试的小软件居然有点模样了，AgileIM 如今已经支持的主要功能：
(1)文字聊天
(2)文件传输
(3)视频/音频聊天
(4)基于Udp的NAT/NAPT 穿透
IM类型的软件所需要的框架的能力大致也在这里了，现在把AgileIM的源码开放，希望能对后来者有些许帮助，当然，服务端通信框架的源码时不开放的。即使如此，你也可以从AgileIM中学到：
（1）如何与服务器进行通信、客户端基于Tcp的通信框架的特点
（2）如何组织通信消息/协议
（3）如何进行视频/音频捕捉
（4）多线程
（5）如何使用ESFramework框架的ESFramework.Network.Tcp.Passive空间
AgileIM基于ESFramework构建，使用VS2005开发，代码并不是很多（因为很多能力已经在ESFramework框架中实现），组件装配/IOC使用了Spring框架，数据访问层使用了XCodeFactory。
测试登录帐号: 100000 ,密码: 111111
110000 ,密码: 111111
111000 ,密码: -AgileIM is a similar QQ, MSN chat software, the software is I "inadvertently inserted Liu", AgileIM the "predecessor" was originally just to test my communications framework ESFramework communications capacity Unexpectedly, a two weeks after the tests were 000 for a small software actually a bit like, AgileIM now has support for the main functions : (1) Text-Chat (2), file transfer (3) video/audio chat (4) Based on the Udp NAT/NAP T-penetrating types of IM software required for the ability of the framework is here broadly, now AgileIM of open-source, in hopes of later were shown some help, of course, server communications source framework is not open. Even so, you can learn from AgileIM : (1) how to communicate with the server. Based on DEC client communications fra
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Size: 777216 |
Author: 王懿璞 |
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Description: 基于UDP网络传输协议，可用于视频，音频等多媒体信息的传输。-UDP-based network transmission protocol can be used for video, audio and other multimedia information transmission.
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Size: 4096 |
Author: Hiray |
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Description: C语言编写的UDP简单文件传输协议TFTP，含客户端和服务端
默认的下载路径为E:/up/，文件将被下载到E:/dn/下；默认的上传路径为E:/dn/,文件将被上传到E:/up/下，也可以通过改变源码更改下载上传路径
不仅支持简单的文本传输，也支持视频音频压缩以及大文件的传输，且传输速度快-C language UDP Simple File Transfer Protocol TFTP, including client and server to download the default path for the E:/up /, files will be downloaded to the E:/dn/under the default upload path for the E:/dn /, files will be uploaded to E:/up/under, you can change by changing the source code download upload path not only support the simple text transmission, also supports video and audio compression and transmission of large files, and transmission speed
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Size: 5120 |
Author: 朱昶炜 |
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Description: First of all, the Applet-phone is a SIP User-Agent
with audio and text messaging capabilities. But i s also embedded in
an applet where you can use it as usual if your firewall allow you to
use the UDP transport. If not, you can use TCP to carry your voice in a
voice messaging fashion. Your voice is recorded and encoded locally
then sent as a message, decoded and played back by your peer. Voice
messaging allow you to go through any firewall, so that you can
continue to chat and talk with your friends from an applet !
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Size: 2630656 |
Author: 康佳 |
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Description: 利用TCP/IP协议完成各种文本音频视频文件传输-The use of TCP/IP protocol to complete a variety of text audio video file transfer
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Size: 869376 |
Author: 李丽 |
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Description: 一个用Java编写的网络聊天软件包，里边含有源程序和所用到的音频和图片等文件-The preparation of a network using Java chat software packages, and the inside contains the source code used in audio and picture files, etc.
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Size: 217088 |
Author: Jimmy |
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Description: < Visual C++ 视频 音频实践工程案例精选>>
国内音视频处理经典教材,四维科技,刘依玮著人民邮电出版社,-<Visual C++ Video audio engineering practice cases Featured>> home audio and video teaching materials dealing with the classic, four-dimensional technology, according to Liu Wei of Posts
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Size: 32160768 |
Author: jiayongchang |
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Description: The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and provide a means for choosing delivery
mechanisms based upon RTP (RFC 1889).
-The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP (RFC 1889).
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Size: 8192 |
Author: 周一 |
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Description: 基于ARM9(Wince)的网络音视频通信-Based on the ARM9 (Wince) audio and video communications network
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Size: 689152 |
Author: mahuilin |
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Description: 局域网音频广播软件，使用UDP协议，并在元基础上就音频格式使用GSM算法压缩，减少了网络流量。-LAN audio broadcasting software, the use of UDP protocol, and in yuan on basic audio formats used on the GSM compression algorithm, reducing network traffic.
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Size: 210944 |
Author: rf |
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Description: 网络多媒体通信
1、编制一个网络多媒通信软件，实现：
在发送端采集话筒声音，通过网络实时传输到接收端，并在接收端播放出来。
2、通过使用TCP、UDP、变更分组大小来对比收发端声音同步情况及播放质量。
本实验技术不同于课上所讲的回调函数，利用了MFC的消息处理机制，用消息处理函数替代了回调函数，但整个流程是一样的。本程序采用C/S模式，其中Server端为项目PhoneToFile，Client端为项目Client，Server端的功能为采集声音数据并发送给客户端，Client端将收到的声音数据播放。在测试中只需在Server端打开Server程序并播放音乐或用话筒录音，在Cliet端打开Client程序，用耳机就可以听到音乐或录音。-Internet Multimedia Communications
1, the preparation of a network of multi-media communications software to achieve:
In the transmitter microphone capture sound, real-time transmission through the network to the receiving side, and playing out at the receiving end.
2, using TCP, UDP, change the packet size to compare the situation and simultaneously send and receive-side audio playback quality.
The experimental technique is different from the class talked about the callback function, use of MFC s message handling mechanism is replaced by the message handler callback function, but the whole process is the same. This program uses C/S mode, in which Server-side for the project PhoneToFile, Client-side for the project Client, Server-side functionality for the capture audio data and sent to the client, Client-side will receive the voice data playback. In the test, simply open the Server in the Server-side program and play music or microphone recording, open the Client program
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Size: 66560 |
Author: zym |
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Description: LanMsg已更名为ourmsg，当前版本为1.1,修正了1.0版本中UDP组件的BUG，调试通过视频编解码，音视频即时消息软件源码。当前版本音频采用G729A算法编解码，视频采用微软MPEG4编解码。视频编解码器在Update\mpg4vki文件夹中，发布客户端打包时需要将mpg4vki一同发布，否则客户端视频将不能正常使用。-LanMsg has been renamed ourmsg, the current version 1.1, revised 1.0 version of UDP component BUG, debugging through the video codec, audio and video instant messaging software source code. The current version of the audio algorithms using G729A codec, video powered by Microsoft MPEG4 codec. Video codec in the Update \ mpg4vki folder, released when the client needs to be packaged together mpg4vki release, otherwise the client-side video will not work properly.
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Size: 7884800 |
Author: tangcl |
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Description: 用C实现的录音和播放程序，udp的client负责录音，udp的server负责同步播放。在录音数据传输前，采用g711u方式进行编码，在接收端解码后再进行播放。代码清晰，注释详细。-Implemented using C recording and playback programs, udp The client is responsible for recording, udp broadcast of the server is responsible for synchronization. In the recording, before the data transmission, using g711u approach to coding and then decoding at the receiving end for playback. Code is clear and detailed comments.
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Size: 56320 |
Author: 孙涛 |
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Description: gm81xx tcp udp audio streaming server
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Size: 6144 |
Author: antoxyz |
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Description: 音频采集得到PCM数据，然后进行AMR编码，再封RTP包，加DTP头，最后送UDP Socket发送，Visual C++ 6.0工程源码-PCM audio data to be collected and then AMR coding, and then sealed RTP packets, plus DTP head, the last to send UDP Socket send, Visual C++ 6.0 Project Source
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Size: 133120 |
Author: 何朝阳 |
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Description: 使用c++编写了对点 语音聊天软件，实现两台主机之间得语音通信，首先通过声音采集，然后利用ACM接口压缩音频数据，将数据发送到目标主机，主机接收数据后，解压缩音频数据，并播放，以此实现了点对点得语音通信。利用TCP/IP协议来确认连接状态（TCP/IP协议是可靠的协议，TCP负责发现传输的问题，一有问题就发出信号，要求重新传输，直到所有数据安全正确地传输到目的地，），UDP协议（UDP是一个无连接协议，传输数据之前源端和终端不建立连接，当它想传送时就简单地去抓取来自应用程序的数据，并尽可能快地把它扔到网络上）来实现数据的发送和接收。ACM WAVEX -Using c++ write software for peer to peer voice chat, to achieve voice communication between two host computers were first collected by the sound, and then use ACM compressed audio data interfaces, data sent to the target host, the host receives data, extract audio data, and play, to assert the point was voice communications. Using TCP/IP protocol to confirm the connection status (TCP/IP protocol is a reliable protocol, TCP is responsible for the problems found in transmission, a signal problem on the demand for re-transmission until all the data security and correctly transferred to the destination), UDP protocol (UDP is a connectionless protocol, transmission of data source and the terminal before the connection is not established, when it simply when you want to transfer to crawl from the application of data and throw it as quickly as possible on the network ) to achieve data transmission and reception.
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Size: 2726912 |
Author: 里小号 |
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Description: udp/ip stack for just streaming the data over IP video or audio vhdl code to run in vhdl
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Size: 180224 |
Author: prasad |
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Description: emiplib库，可以实现pc到开发板arm的网络语音通讯-emiplib arm rtp udp audio speex
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Size: 1637376 |
Author: zj |
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