gecko/media/webrtc/trunk
Ehsan Akhgari 7fb2d3d770 Remove media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf, which seems to have randomly been added in bug 987979
DONTBUILD

--HG--
extra : amend_source : 1b3ff1e9a5b0eb721f5209e21187f433c24294fa
2014-12-23 21:25:50 -05:00
..
base
build Bug 1072296 - make webrtc only define WINVER and _WIN32_WINNT if we're not building for mozilla; r=mshal 2014-10-27 15:50:43 -04:00
chromium_deps
google_apis/build
net
supplement
testing
third_party/opus
tools
webrtc Remove media/webrtc/trunk/webrtc/tools/e2e_quality/audio/perf, which seems to have randomly been added in bug 987979 2014-12-23 21:25:50 -05:00
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