glaznev@webrtc.org e3fccd4268 Merge changes from internal repo to AppRTCDemo. ... - Add a setting option to disable outgoing video in a call. - Add an option to select audio codec. - Add an option to specify audio bitrate for Opus codec. - Plus add an option to select H.264 as default video codec. R=wzh@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42449004 Cr-Commit-Position: refs/heads/master@{#8468} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8468 4adac7df-926f-26a2-2b94-8c16560cd09d		2015-02-24 00:54:00 +00:00
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android	Merge changes from internal repo to AppRTCDemo.	2015-02-24 00:54:00 +00:00
androidtests	Merge fixes and changed for Android AppRTCDemo from internal repo.	2015-02-10 23:05:04 +00:00
objc	Update GAE API paths for join/leave.	2015-01-27 21:35:16 +00:00
peerconnection	Don't call g_thread_init on glib >=2.31.0	2015-02-17 21:23:13 +00:00
relayserver	Update libjingle license statements at top of talk files for consistency	2015-01-20 21:36:13 +00:00
stunserver	Update libjingle license statements at top of talk files for consistency	2015-01-20 21:36:13 +00:00
turnserver	Update libjingle license statements at top of talk files for consistency	2015-01-20 21:36:13 +00:00
OWNERS	Add new OWNERS file to talk/examples.	2014-08-07 20:01:34 +00:00