搜索资源列表
AndroidrRTC_WebScoke
- 实现手机端web_rtc和后台的音视频通信.这个是源码修改后的项目.java端的代码可以在网上找node.js开发的webrtc源码. 该手机端是webscoket和http进行通信的-Achieve mobile terminal web_rtc and background audio and video communications. Background code can be found on the Internet web
vad
- webrtc extracted vad algorithm. Witch is written by c++。
fft
- Changes: Trivial type modifications by the WebRTC authors.
AEC
- C/C++ 的AEC 算法 取自于google 的webrtc 开源项目-C/C++ AEC form google webrtc project
audio_device
- webrtc modules audio device include audio device defines.h Kernel Device Driver for linux v2.13.6.
AgoraMediaSDK-1.0.1-windows
- webrtc sdk接口,里面有比较标准的例子,属于声网-webrtc sdk
ice4j
- webrtc ice 封装,协议为了提供一个强大的机制,它允许基于提供/回答协议SIP和XMPP遍历NATs等。-The Interactive Connectivity Establishment (ICE) protocol combines various NAT traversal utilities such as the STUN and TURN protocols in order to offer a powerful
webrtc_vad
- WebRTC中提取出来的vad检测代码,目前读取全部为1。希望各位高手能够修正。-WEBrtc vad dection,now has a problem,read all data is 1,hope others can resolve it.
echo
- webrtc的回声消除模块例子,vc2008用的!-Webrtc echo cancellation module, VC used!
testAudio
- webrtc的回声消除,这个是源码要编译的,vc用的-Webrtc echo cancellation, this is the source code to be compiled, VC used
gyp_ninja
- Webrtc Build System (gyp/ninja) Basic Program to Start with
AEClib
- AEC库,回声抑制库,以至于webRTC源码。使用VS2013编译通过,头文件echo_cancelletion.h-AEC library, library echo suppression that webRTC source. Using VS2013 compiler, header files echo_cancelletion.h
aec-master
- webrtc 的回声抵消(aec、aecm)算法主要包括以下几个重要模块:1.回声时延估计 2.NLMS(归一化最小均方自适应算法) 3.NLP(非线性滤波) 4.CNG(舒适噪声产生),一般经典aec算法还应包括双端检测(DT)。考虑到webrtc使用的NLMS、NLP和CNG都属于经典算法范畴,故只做简略介绍,本文重点介绍webrtc的回声时延估计算法,这也是webrtc回声抵消算法区别一般算法(如视频会议中的算法)比较有特色的地方
WebRTC-Experiment-master
- Web RTC,基于Firfox和Chrome浏览器可以实现语音和视频通话的源程序-Web RTC, a tool to have audio or video call by Firefox or Chrome Browser.
VAD
- 附件内容为从Webrtc程序中抠出来的VAD检测算法-Webrtc VAD code
webrtc-video-chat-master
- Video Chat with web rtc
webrtc_vad
- VAD语音活动检测,在linux下编译,带测试程序,本vad库由webrtc中单独抽取出来编译而成。-VAD voice activity detection, under the Linux compiler, with testing procedures, the VAD library by webrtc extracted a separate.
webrtc_ns
- webrtc噪声抑制算法,个人做了算法改进包含测试工程-noise suppression algorithm in webrtc,include test project.
jssip-web-client
- jssip 一个基于webrtc的web sip电话功能。可以连接websocket-JsSIP: The Javascr ipt SIP Library Runs in the browser and Node.js SIP over WebSocket (use real SIP in your web apps) Audio/video calls (WebRTC) and instant
apprtc-master
- webrtc demo服务器。提供信令服务,穿透服务,房间服务。-webrtc demo server