gecko-dev/media/webrtc/trunk
2013-06-08 12:47:31 +02:00
..
base Bug 797671: Import Webrtc.org code from stable branch 3.12 (rev 2820) rs=jesup 2012-10-04 12:09:31 -04:00
build Bug 807492 - reland after fixing a typo r=try-green 2013-02-24 15:34:00 +01:00
chromium_deps Bug 797671: Import Webrtc.org code from stable branch 3.12 (rev 2820) rs=jesup 2012-10-04 12:09:31 -04:00
google_apis/build Bug 830247: Update media/webrtc/trunk except /webrtc (tools, etc) r=derf 2013-02-09 23:16:09 -05:00
net Bug 797671: Import Webrtc.org code from stable branch 3.12 (rev 2820) rs=jesup 2012-10-04 12:09:31 -04:00
supplement Bug 830247: rollup of changes to media/webrtc/trunk, and backouts of some temp patches r=ted,derf 2013-02-09 23:16:10 -05:00
testing Bug 807492 Part 3 - Backport chunk of upstream gtest r629 to fix <tuple> detection on BSDs with old libstdc++, not breaking it on MacOSX r=upstream 2013-06-13 08:41:49 +02:00
third_party Bug 878446 - Disable libyuv asm without SSSE3 as well. r=jesup 2013-06-08 12:44:26 +02:00
tools Bug 775939 - Fix gyp Makefile include error on msvc (change topsrcdir, srcdir and VPATH to absolute path). r=ted 2012-12-30 23:19:38 +09:00
webrtc Bug 844818 - Make WebRTC aware of --enable-alsa/--enable-pulseaudio. r=jesup,ted 2013-06-08 12:47:31 +02:00
DEPS Bug 830247: Update media/webrtc/trunk except /webrtc (tools, etc) r=derf 2013-02-09 23:16:09 -05:00
dummy_file.txt Bug 797671: Import Webrtc.org code from stable branch 3.12 (rev 2820) rs=jesup 2012-10-04 12:09:31 -04:00
Makefile.old Bug 830247: Update media/webrtc/trunk except /webrtc (tools, etc) r=derf 2013-02-09 23:16:09 -05:00
OWNERS
peerconnection_client.target.mk Bug 830247: small changes resulting from Try build r=ted rs=me 2013-02-09 23:16:10 -05:00
peerconnection.gyp Bug 830247: rollup of changes to media/webrtc/trunk, and backouts of some temp patches r=ted,derf 2013-02-09 23:16:10 -05:00
peerconnection.Makefile Bug 797671: Import Webrtc.org code from stable branch 3.12 (rev 2820) rs=jesup 2012-10-04 12:09:31 -04:00
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