external/webrtc.git -

Age	Commit message (Expand)	Author
2016-01-14	Add ramp-up tests for transport sequence number with and w/o audio.	Stefan Holmer
2015-12-10	Replace EventWrapper in video/, test/ and call/.	Peter Boström
2015-12-09	Nuke TickTime::UseFakeClock.	Peter Boström
2015-11-26	Clean up PlatformThread.	Peter Boström
2015-11-23	Move ThreadWrapper to ProcessThread in base.	pbos
2015-10-28	system_wrappers: rename interface -> include	Henrik Kjellander
2015-10-27	Call OnSentPacket for all packets sent in the test framework.	stefan
2015-10-02	Add a PacketOptions struct to webrtc::Transport.	stefan
2015-05-01	Use rtc::CriticalSection in webrtc/video/.	Peter Boström
2015-03-22	Re-land 8810 "- Add a SetPriority method to ThreadWr..."	tommi@webrtc.org
2015-03-22	Revert 8810 "- Add a SetPriority method to ThreadWrapper"	tommi@webrtc.org
2015-03-20	- Add a SetPriority method to ThreadWrapper	tommi@webrtc.org
2015-03-13	Remove thread id from ThreadWrapper::Start().	pbos@webrtc.org
2015-02-13	Remove EventWrapper::Reset().	pbos@webrtc.org
2014-12-15	Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.	pkasting@chromium.org
2014-02-26	Add SetConfig method to FakeNetworkPipe and to DirectTransport	henrik.lundin@webrtc.org
2013-12-18	Integrate fake_network_pipe into direct_transport.	stefan@webrtc.org
2013-11-20	Rename newapi::Transport::SendRTP()->SendRtp().	pbos@webrtc.org
2013-11-18	Hook up audio/video sync to Call.	stefan@webrtc.org
2013-11-15	Improve Call tests for RTX.	stefan@webrtc.org
2013-10-28	Separate Call API/build files from video_engine/.	pbos@webrtc.org