webrtc

History

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

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