gecko-dev/media/webrtc/trunk
Jan Beich 6f8049a310 Bug 910875 - Add missing ifdefs to make audio_device work on BSDs. r=jesup 2013-08-30 22:13:55 +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 872127 - Part 2: Replace mozilla/StandardInteger.h with stdint.h; r=Waldo,ted 2013-07-30 10:25:31 -04:00
tools Bug 907473 - Handle generator_flags gracefully in gyp. r=gps 2013-08-21 09:37:45 +09:00
webrtc Bug 910875 - Add missing ifdefs to make audio_device work on BSDs. r=jesup 2013-08-30 22:13:55 +02:00
DEPS Bug 830247: Update media/webrtc/trunk except /webrtc (tools, etc) r=derf 2013-02-09 23:16:09 -05:00
Makefile.old Bug 830247: Update media/webrtc/trunk except /webrtc (tools, etc) r=derf 2013-02-09 23:16:09 -05:00
OWNERS
README
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
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
peerconnection.gyp Bug 901583: Reapply mozilla patches on top of webrtc.org 3.34, use NEON detection rs=jesup 2013-08-30 02:08:57 -04:00
peerconnection_client.target.mk Bug 830247: small changes resulting from Try build r=ted rs=me 2013-02-09 23:16:10 -05:00

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