This is an Android Studio reference project for WebRTC based video and voice calling app.
- date:2022/02/09
- version: branch-heads/4855
gn configuration is：
gn gen out/android/release/arm --args='is_debug=false target_os="android" target_cpu="arm" rtc_use_h264=true ffmpeg_branding="Chrome" proprietary_codecs=true '
gn gen out/android/release/arm64 --args='is_debug=false target_os="android" target_cpu="arm64" rtc_use_h264=true ffmpeg_branding="Chrome" proprietary_codecs=true '
gn gen out/android/release/x64 --args='is_debug=false target_os="android" target_cpu="x64" rtc_use_h264=true ffmpeg_branding="Chrome" proprietary_codecs=true '
gn gen out/android/release/x86 --args='is_debug=false target_os="android" target_cpu="x86" rtc_use_h264=true ffmpeg_branding="Chrome" proprietary_codecs=true 'autoninja -C out/android/release/xxx sdk/android:libjingle_peerconnection_so