webrtc源码获取及编译

1、没安装svn就先装一个可以运行的版本

2、下载并配置depot_tools

    　建立一个存放depot_tools的目录，command进入该目录

   　 svn co http://src.chromium.org/svn/trunk/tools/depot_tools

    　下载后把depot_tools目录 手动添加到系统环境变量PATH中

3、建一个webrtc目录　eg. d:\webrtc\

4、打开cmd,进入d:\webrtc

        d:\webrtc> gclient

        这时会提示安装一些工具，如git, svn, python 不管他，它会一直在后台下载并最后自动配置好，根据各自网络不同时间不同

　　d:\webrtc> gclient

　　没有提示安装什么了，就可以了，这时会提示gclient的一个参数说明

　

5、创建 .gclient 文件

　　d:\webrtc> gclient config http://webrtc.googlecode.com/svn/trunk

　　这时在d:\webrtc\下多了一个文件

6、下载webrtc代码及工作环境

　　d:\webrtc>gclient sync

　　生成　d:\webrtc\trunk　文件夹并慢慢下载文件，可能中途会断掉，多重复几次就可以了

7、生成

　　d:\webrtc>gclient runhooks

    

  NOTE: The gclient sync command fetches dependencies and generates native build files for your environment using gyp (Windows: ninja/Visual Studio, OS X: ninja/XCode, Linux: ninja/make, Android: ninja). Ninja is the default build system for all platforms.

Select build system

You can directly specify which build system to use. This can be done if you don't want to use ninja. Set the GYP_GENERATORS environment variable to the string:

make for Makefiles

msvs for Visual Studio

xcode for Xcode

8、　如果在windows上想生成Visual Studio 工程，在系统环境变量中加上 GYP_GENERATORS 值为msvs for Visual Studio后，再次　gclient runhooks　即可。

　　这样后就可以生成Visual Studio工程了。

 

webrtc 的原文：

http://www.webrtc.org/reference/getting-started

Before you start

First, be sure to install the prerequisite software.

The currently supported platforms are Windows, Mac OS X, Linux and Android.

Getting the code and building

Create a working directory, enter it, and run:

$ gclient config http://webrtc.googlecode.com/svn/trunk

Android specific steps. *

Select build system (optional for all OSs except Android where ninja is mandatory). *

$ gclient sync --force

Starred (*) items are described in their own section below and should be performed, if at all, in place.

Notes

If you're a committer, substitute https for http.

On Windows, use gclient.bat instead (or prefix the commands by invoking python).

Android requires that you build on a Linux machine.

The gclient sync command fetches dependencies and generates native build files for your environment using gyp ( Windows: ninja/ Visual Studio, OS X: ninja/XCode, Linux: ninja/ make , Android: ninja). Ninja is the default build system for all platforms. It is possible to just generate new build files by calling:

$ gclient runhooks --force

Android Specific Steps

If building for Android these steps should be inlined above.

$ echo "target_os = ['android', 'unix']" >> .gclient

$ gclient sync --nohooks

$ source ./build/android/envsetup.sh

$ JAVA_HOME=<location of Java SE 6 - JDK>

Select build system

You can directly specify which build system to use. This can be done if you don't want to use ninja. Set the GYP_GENERATORS environment variable to the string:

make for Makefiles

msvs for Visual Studio

xcode for Xcode

Note, when the build environment is set to generate Visual Studio project files, gyp will by default, generate a project for the latest version of Visual Studio installed on your computer. It is possible to specify the desired Visual Studio version as described below:

Set environment variable GYP_MSVS_VERSION=<version> before runhooks or manually run the following gyp command from the trunk/ directory (this replaces gclient runhooks ):

$ build/gyp_chromium --depth=. -G msvs_version=<version> all.gyp

<version> is on the form YYYY .

Building

Binaries are by default (i.e. when building with ninja) generated in out/Debug/ and out/Release for debug and release builds respectively.

With ninja

$ cd trunk

Debug:

$ ninja -C out/Debug

Release:

$ ninja -C out/Release

With Visual Studio

Use Visual Studio to open and build the trunk/all.sln solution file.

Example Applications

WebRTC contains several example applications which can be found under trunk/webrtc/examples and trunk/talk/examples. Higher level applications are listed first.

AppRTCDemo (Android application using WebRTC Native APIs via JNI)

The JNI wrapper is documented here. AppRTCDemo is documented here.

Peerconnection (Application using WebRTC Native APIs)

Peerconnection consist of two applications. A server application, peerconnection_server , and a client application, peerconnection_client . Note that we currently don't support peerconnection_client for Mac and Android.

The client application has simple voice and video capabilities. The server enables client applications to initiate a call between clients by managing signaling messages generated by the clients.

Setting up P2P calls between peerconnection_clients

Start peerconnection_server . You should see the following message indicating that it is running:

Server listening on port 8888

Start any number of peerconnection_client s and connect them to the server. The client UI consists of a few parts:

Connecting to a server: when the application is started you must specify which machine (IP-address) the server application is running on. Once that is done you can press "Connect" or the return button.

Select a peer: once successfully connected to a server you can connect to a peer by double clicking or select+press return on a peer's name.

Video chat: when a peer has been successfully connected to, a Video chat will be displayed in full window.

Ending chat session: press Esc. You will now be back to selecting a peer.

Ending connection: press Esc and you will now be able to select  which server to connect to.

Testing peerconnection_server

Start an instance of peerconnection_server application.

Open trunk/talk/examples/peerconnection/server/server_test.html in your browser. Click connect. Observe that the peerconnection_server announces your connection. Open one more tab using the same page. Connect it too (with a different name). It is now possible to exchange messages between the connected peers.

Call (Application that establishes a call using libjingle)

Call uses xmpp (as opposed to SDP used by WebRTC) to allow you to login using your gmail account and make audio/video calls with your gmail friends . It is built on top of libjingle to provide this functionality.

Further, y ou can specify input and output RTP dump for voice and video. It provides two samples of input RTP dump: voice.rtpdump which contains a stream of single channel, 16Khz voice encoded with G722, and video.rtpdump which contains a 320x240 video encoded with H264 AVC at 30 frames per second. The provided samples will inter-operate with Google Talk Video. If you use other input RTP dump, you may need to change the codecs in call_main.cc (lines 215 - 222).

WebRTCDemo (Android application using media engines)

This app does not use WebRTC native APIs. It can send and receive media streams if manually connected to another WebRTCDemo that is directly accessible (e.g. firewalls might prevent you from establishing a connection). Further it allows setting, enabling and disabling audio and video processing functionality (e.g. echo cancellation, NACK, audio codec and video codec).

Relay server (specialized server application that can be used with Call)

Target name relayserver . Relays traffic when a direct peer-to-peer connection can't be established.

Stun server

Target name stunserver . Implements the STUN protocol for Session Traversal Utilities for NAT as documented in rfc5389.

Turn server

Target name turnserver . In active development to reach compatibility with rfc5766.