https://webrtc.org/native-code/development/

趁着能访问先备份一下.

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The currently supported platforms are Windows, Mac OS X, Linux, Android and iOS. See the Android and iOS pages for build instructions and example applications specific to these mobile platforms.

Before You Start

First, be sure to install the prerequisite software.

Getting the Code

For desktop development:

Create a working directory, enter it, and run fetch webrtc:

mkdir webrtc-checkout

cd webrtc-checkout

fetch--nohookswebrtc

gclient sync

NOTICE: During your first sync, you’ll have to accept the license agreement of the Google Play Services SDK.

The checkout size is large due the use of the Chromium build toolchain and many dependencies. Estimated size:

Linux: 6.4 GB.

Linux (with Android): 16 GB (of which ~8 GB is Android SDK+NDK images).

Mac (with iOS support): 5.6GB

Optionally you can specify how new branches should be tracked:

git config branch.autosetupmerge always

git config branch.autosetuprebase always

Alternatively, you can create new local branches like this (recommended):

cd src

git checkout master

git new-branch your-branch-name

See Android and iOS pages for separate instructions.

NOTICE: if you get Remote: Daily bandwidth rate limit exceeded for , make sure you’re logged in. The quota is much larger for logged in users.

Updating the Code

Update your current branch with:

git checkout master

git pull origin master

gclient sync

git checkout my-branch

git merge master

Building

Ninja is the default build system for all platforms.

See Android and iOS for build instructions specific to those platforms.

Generating Ninja project files

Ninja project files are generated using GN. They’re put in a directory of your choice, like out/Debug or out/Release, but you can use any directory for keeping multiple configurations handy.

To generate project files using the defaults (Debug build), run (standing in the src/ directory of your checkout):

gn gen out/Default

To generate ninja project files for a Release build instead:

gn gen out/Default--args='is_debug=false'

To clean all build artifacts in a directory but leave the current GN configuration untouched (stored in the args.gn file), do:

gn clean out/Default

See the GN documentation for all available options. There are also more platform specific tips on the Android and iOS pages.

Compiling

When you have Ninja project files generated (see previous section), compile (standing in src/) using:

For Ninja project files generated in out/Default:

ninja-Cout/Default

Using Another Build System

Other build systems are not supported (and may fail), such as Visual Studio on Windows or Xcode on OSX. GN supports a hybrid approach of using Ninja for building, but Visual Studio/Xcode for editing and driving compilation.

To generate IDE project files, pass the --ide flag to the GN command. See the GN reference for more details on the supported IDEs.

Working with Release Branches

To see available release branches, run:

git branch-r

To create a local branch tracking a remote release branch (in this example, the 43 branch):

git checkout-bmy_branch refs/remotes/branch-heads/43

gclient sync

NOTICE: depot_tools are not tracked with your checkout, so it’s possible gclient sync will break on sufficiently old branches. In that case, you can try using an older depot_tools:

which gclient

# cd to depot_tools dir

# edit update_depot_tools; add an exit command at the top of the file

git log # find a hash close to the date when the branch happened

git checkout <hash>

cd~/dev/webrtc/src

gclient sync

# When done, go back to depot_tools, git reset --hard, run gclient again and

# verify the current branch becomes REMOTE:origin/master

The above is untested and unsupported, but it might help.

Commit log for the branch: https://webrtc.googlesource.com/src/+log/branch-heads/43

To browse it: https://webrtc.googlesource.com/src/+/branch-heads/43

For more details, read Chromium’s Working with Branches and Working with Release Branches pages.

Contributing Patches

Please see Contributing Fixes for information on how to run git cl upload, getting your patch reviewed, and getting it submitted.

Chromium Committers

Many WebRTC committers are also Chromium committers. To make sure to use the right account for pushing commits to WebRTC, use the user.email Git config setting. The recommended way is to have the chromium.org account set globally as described at the depot tools setup page and then set user.emaillocally for the WebRTC repos using (change to your webrtc.org address):

cd/path/to/webrtc/src

git config user.email yourname@webrtc.org

Example Applications

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

Peerconnection

Peerconnection consist of two applications using the WebRTC Native APIs:

A server application, with target name peerconnection_server

A client application, with target name peerconnection_client (not currently supported on Mac/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_clients 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 (by 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 src/webrtc/examples/peerconnection/server/server_test.html in your browser. Click Connect. Observe that the peerconnection_serverannounces 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 App

Target name call (currently disabled). An 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, you 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 interoperate 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).

Relay Server

Target name relayserver. Relays traffic when a direct peer-to-peer connection can’t be established. Can be used with the call application above.

STUN Server

Target name stunserver. Implements the STUN protocol for Session Traversal Utilities for NAT as documented in RFC 5389.

TURN Server

Target name turnserver. In active development to reach compatibility with RFC 5766.