WEBRTC Compilation Details--record + reprintOriginal address: http://blog.csdn.net/temotemo/article/details/7056581WEBRTC compilingMy environment:Operating system: XP SP3VS 2013Tools required before compiling the source codeGet the source code tool:1, first need to install the source of the tool SVN (Project code version management tools, Google also use this)TortoiseSVN 1.6.12http://sourceforge.net/projects/tortoisesvn/2. Download and install Msysgit
opportunity to use, and later do OTT voice (app voice) used in the WEBRTC 3A algorithm. After doing the audio development on the Android mobile platform, I used the Neteq on the WEBRTC, but used the earlier C language version, not the C + + version, and only involved the DSP module in Neteq (Neteq has two modules, MCU (Micro control unit , Micro Control Unit) an
1,HTTP://WWW.WEBRTC.ORG/WEBRTC official website, god Horse compilation, God horse download, the solution here is the most authoritative.---------------------------------2,HTTPS://CODE.GOOGLE.COM/P/WEBRTC/WEBRTC Source download location, you can also pay attention to the latest changes anywhere.----------------------------------3,https://webrtchacks.com/is an arti
reproduced in the original: http://www.cnblogs.com/lidabo/p/6842765.html thank you very much.
"WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is a technology that supports web browsers for real-time voice conversations or video conversations, and is Google's $68.2 million acquisition of Global in 2010 IP Solutions A technology that Google has made available to the company. ”
"WebRTC, a name derived from the abbreviation of Web real-time communication (Web real-time communication), is a technology that supports web browsers for real-time voice conversations or video conversations, and is Google's $68.2 million acquisition of global IP in 2010 Solutions company to get a technology, Google Open source of this technology! ”Just for the more than 60 million dollar technology, it is worth studying!
/video_x11_channel.h: 20: 33: Fatal error: X11/extensions/XShm. h: the file or directory compilation is not interrupted.Make: *** [out/Debug/obj.tar get/video_render_module/webrtc/modules/video_render/linux/video_x11_channel.o] Error 1
Install a libraryRoot @ zhangsj-mac:/home/webrtc_svn/trunk # apt-get install libxext-dev
Finally, Make is successful.
July 2014 update: Ubuntu 14.04 compiled WebRTC For
Android App templateCrosswalk Android distribution contains an application template that can be used as a wrapper for an HTML5 application. It also includes a script to convert the wrapped HTML5 application into an installable Android apk file.Let the crosswalk Android:
Download the version you want on the do
WebRTC, a name derived from the abbreviation of Web real-time communication ( English:Web Real-time communication), is an API that supports Web browsers for real-time voice conversations or video conversations. It was open source on June 1, 2011 and was included in the World Wide Web Consortium's recommended standard for Google, Mozilla and opera support [ 1] [2] [3]. Http://baike.baidu.com/link?url=G9wblLo409MIqXQW1XDplFtdKgyol5_LXG8N4cxSYQzXuqc1blHy
WEBRTC rtp/rtcp Protocol family2017-02-22 20:15 Reading (144) Comments (0) WebRTC congestion control based on GCC (bottom)2017-02-22 15:44 Reading (108) Comments (0) WebRTC congestion control based on GCC (upper)2017-02-22 11:37 Review (0) WebRTC video receive buffer based on Kalmanfilter delay model2017-02-22 11:2
WebRTC's Past Life
This article is translated by RWEBRTC
WebRTC 技术是激烈的开放的 Web 战争中一大突破。-Brendan Eich, inventor of JavaScriptNo plug-in real-time communicationImagine mobile phones, TVs, and computers communicating through a unified platform. Just imagine, it's easy to add video chat and peer-to data sharing to your site. This is the vision of WebRTC technology.Want to have a try? WEBRT c is ava
Recently doing a mobile end with mobile, web-side text, video, voice chat features. Text chat using WebSocket, a lot of information on the Internet, there is no difficulty. But in the video, voice chat encountered a small difficulty. have been looking for some of the SDK to quickly develop, such as Opentok, cloud communications, etc., but the project is used in the intranet, these SDKs must be used in an external network, you need to obtain signaling on their servers. Later, I will try to use
In fact, as early as June 2 ago, a friend who worked at Google told me this information. I also got all the source code from WebRTC for the first time, but since my recent work was really busy, this information was not immediately reproduced here. Now, I want to keep an eye on the multimedia applications, learn the technologies in the WebRTC Framework earlier, and use them in actual projects.
Google today
[Transfer to]WEBRTC Learning: Deploying Stun and turn serversHttp://www.cnblogs.com/lingdhox/p/4209659.htmlThe WEBRTC-to-peer penetration part is implemented by Libjingle.The sequence of steps is probably this:1. Try direct Connect.2. Penetrate through the stun server3. Cannot penetrate through the turn server relay.Stun server is relatively simple. There are also many public stun servers available for test
, While the Android audio system latency is mostly above 100ms, so it is necessary to increase the length of the AEC-PC filter and ensure its operational efficiency is the focus of optimization) 3 other module optimization (such as jitter buffer, etc.).
4. After the text of the source list is outdated, because I do not currently support the separate compilation of these modules, I do not update the list, if there is an independent compiler requirement
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.When using WEBRTC Base library to develop Android,ios native application, because the mobile terminal is not like PC side, in bandwidth stability, system performance is very big difference, so for mobile device WEBRTC need to do some optimization to improve the call effect,For exam
Google open real-time communication framework WEBRTC source code
In fact, as early as June 2, in Google work friends told me this message, I also first time to get WEBRTC all source code, but because the recent work is busy, not the first time in this reprint this information. Now, I hope to pay attention to the peer of multimedia application, can learn the technology of
The text of this text connection is: http://blog.csdn.net/freewebsys/article/details/47174209 not allowed to reprint without the Bo master.1, Encounter problemsFirst of all, WEBRTC is a very good open source project, it is a company that specializes in this, was acquired by Google and then open source projects.You can quickly build a video chat project, and you can compile it yourself.Https://github.com/pristineio/
Transferred from: http://www.cnblogs.com/fangkm/p/4370492.htmlReprint Please specify source: http://www.cnblogs.com/fangkm/p/4370492.htmlThe previous article simply introduced the next WEBRTC protocol process, which begins with the introduction of frameworks and interfaces.When it comes to frames, instinctively don't know where to start. Once directly from the chromium project on the integration of the source of W
This article original from Http://blog.csdn.net/voipmaker reprint annotated source.WEBRTC provides real-time, web-based audio and video data interoperability, but WEBRTC can also run as a native app on a mobile platform, WEBRTC is a set of media frameworks, implemented in C + +, and officially ported to mobile platforms, including Android,ios, Platform-correspond
WebRTC supports the use of its own codec (limited to native development), audio, video can be. Here the video coding as an example to analyze the corresponding source code in the WebRTC.CreatepeerconnectionfactoryIn Webrtc/api/peerconnectioninterface.h there is a method Createpeerconnectionfactory, the prototype is as follows:inline rtc::scoped_refptrAs you can see, the last four parameters of Createpeercon
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