# Webrtc_Android

**Repository Path**: linving/Webrtc_Android

## Basic Information

- **Project Name**: Webrtc_Android
- **Description**: 对webrtc进行扩展，使其具有支持视频会议的能力
- **Primary Language**: Unknown
- **License**: Not specified
- **Default Branch**: master
- **Homepage**: None
- **GVP Project**: No

## Statistics

- **Stars**: 0
- **Forks**: 1
- **Created**: 2015-10-22
- **Last Updated**: 2020-12-18

## Categories & Tags

**Categories**: Uncategorized

**Tags**: None

## README

This directory contains a sample app for sending and receiving video and audio
on Android. It further lets you enable and disable some call quality
enhancements such as echo cancellation, noise suppression etc.

Prerequisites:
- Make sure gclient is checking out tools necessary to target Android: your
  .gclient file should contain a line like:
  target_os = ['android']
  Make sure to re-run gclient sync after adding this to download the tools.
- Env vars need to be set up to target Android; easiest way to do this is to run
  (from the libjingle trunk directory):
  . ./build/android/envsetup.sh
  Note that this clobbers any previously-set $GYP_DEFINES so it must be done
  before the next item.
- Set up webrtc-related GYP variables:
  export GYP_DEFINES="$GYP_DEFINES java_home=</path/to/JDK>"
- Finally, run "gclient runhooks" to generate Android-targeting .ninja files.

Example of building the app:
cd <path/to/repository>/trunk
ninja -C out/Debug WebRTCDemo

It can then be installed and run on the device:
adb install -r out/Debug/WebRTCDemo-debug.apk