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path: root/video_engine/vie_rtp_rtcp_impl.h
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2014-02-27Adds APIs for reporting pacer queuing delay.jiayl@webrtc.org
BUG=2775 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8959005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19Add RTCP packet type counter (for getting statistics such as sent/received ↵asapersson@webrtc.org
NACK and FIR). Add counter to RTCP sender and RTCP receiver. Add video api GetRtcpPacketTypes(). BUG=2638 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-10Add stats of incoming frame delays for debugging bandwidth estimation.jiayl@webrtc.org
BUG=crbug/338380 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21Added API for enabling/disabling RTCP Receiver Reference Time extension.asapersson@webrtc.org
R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3419005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Interface changes to old api, for use by new api transition.sprang@webrtc.org
BUG=2589 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3209004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5142 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-20Add functions to ViE API to enable/disable the absolute send time header ↵solenberg@webrtc.org
extension. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1487004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4065 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-17Include files from webrtc/.. paths in video_engine/pbos@webrtc.org
BUG=1662 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1492004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4056 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-16Remove SetOverUseDetectorOptions and cleaned ViESharedData.mflodman@webrtc.org
R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1486004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4042 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14Adding a factory to remote bitrate estimator and allow it to be set via config.andresp@webrtc.org
Additionally: - clean api to set remote bitrate estimator mode. - clean api to set over use detector options. R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1448006 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-12Adding a payload type for RTX.mflodman@webrtc.org
BUG=736 TEST=Modified RTP unittests. Review URL: https://webrtc-codereview.appspot.com/1278004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15Adding a receive side API for buffering mode.mikhal@webrtc.org
At the same time, renaming the send side API. Review URL: https://webrtc-codereview.appspot.com/1104004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01Adding a send side API for streamingmikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1070009 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3457 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22Move 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
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