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--HG-- rename : media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.cc => media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.cc rename : media/webrtc/trunk/webrtc/modules/audio_device/android/audio_device_opensles_android.h => media/webrtc/trunk/webrtc/modules/audio_device/audio_device_opensles.h |
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| .. | ||
| base | ||
| build | ||
| chromium_deps | ||
| google_apis/build | ||
| net | ||
| supplement | ||
| testing | ||
| third_party | ||
| tools | ||
| webrtc | ||
| DEPS | ||
| dummy_file.txt | ||
| Makefile.old | ||
| OWNERS | ||
| peerconnection_client.target.mk | ||
| peerconnection.gyp | ||
| peerconnection.Makefile | ||
| README | ||
This folder can be used to pull together the chromium version of webrtc and libjingle, and build the peerconnection sample client and server. This will check out a new repository in which you can build peerconnection_server. Steps: 1) Create a new directory for the new repository (outside the webrtc repo): mkdir peerconnection cd peerconnection 2) gclient config --name trunk http://webrtc.googlecode.com/svn/trunk/peerconnection 3) gclient sync 4) cd trunk 5) make peerconnection_server peerconnection_client