external/chromium_org/third_party/webrtc.git -

Age	Commit message (Collapse)	Author
2013-08-15	Update talk to 50918584.	wu@webrtc.org
	Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16	Revert r4301	tnakamura@webrtc.org
	R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-05	Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the	stefan@webrtc.org
	rtp_rtcp implementation. This refactoring significantly reduces the receive-side RTP parser and receiver complexity, and makes it possible to implement RTX correctly by having two instances of receive-statistics. With this change the dead-or-alive and packet timeout APIs are removed. TEST=trybots, vie_auto_test, voe_auto_test BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1745004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-03	Proper spacing for end-of-namespace comments.	pbos@webrtc.org
	BUG= R=mflodman@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1760006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-22	API to control target delay in NetEq jitter buffer. NetEq maintains the ↵	turaj@webrtc.org
	given delay unless channel conditions require a higher delay. TEST=unit-test, manual, trybots. R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1384005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-11	Adding playout buffer status to the voe video sync	pwestin@webrtc.org
	Review URL: https://webrtc-codereview.appspot.com/1311004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3835 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-12	Implement initial delay. This CL allows clients of VoE to set an initial ↵	turaj@webrtc.org
	delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. TEST=ACM unit test is added, also a manual integration test is writen. Review URL: https://webrtc-codereview.appspot.com/1097009 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22	Move src/ -> webrtc/	andrew@webrtc.org
	TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@2963 4adac7df-926f-26a2-2b94-8c16560cd09d