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path: root/webrtc/modules/rtp_rtcp/interface
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2013-03-15Adding RTX on sourcemikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1190004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-12Remove DTMF detection. Talk team has been in the loop and there is no need forturaj@webrtc.org
DTMF detection at the receiver side. test=voe_auto_test, VoE extended test DTMF Review URL: https://webrtc-codereview.appspot.com/1168004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01Fix mismatch between different NACK list lengths and packet buffers.stefan@webrtc.org
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors. BUG=1289 Review URL: https://webrtc-codereview.appspot.com/1065007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01Break out RemoteBitrateEstimator from RtpRtcp module and make ↵stefan@webrtc.org
RemoteBitrateEstimator::Process trigger new REMB messages. Also make sure RTT is computed independently of whether it's time to send RTCP messages or not. BUG=1298 Review URL: https://webrtc-codereview.appspot.com/1060005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17Break out RtpClock to system_wrappers and make it more generic.stefan@webrtc.org
The goal with this new clock interface is to have something which is used all over WebRTC to make it easier to switch clock implementation depending on where the components are used. This is a first step in that direction. Next steps will be to, step by step, move all modules, video engine and voice engine over to the new interface, effectively deprecating the old clock interfaces. Long-term my vision is that we should be able to deprecate the clock of WebRTC and rely on the user providing the implementation. TEST=vie_auto_test, rtp_rtcp_unittests, trybots Review URL: https://webrtc-codereview.appspot.com/1041004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-14Cleaned up the data path for payload data, made callbacks to rtp_receiver ↵phoglund@webrtc.org
nonoptional. The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch. BUG= TEST=vie & voe_auto_test full runs Review URL: https://webrtc-codereview.appspot.com/1014006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07Reformatted RTPReceiver.phoglund@webrtc.org
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I though that is more risky, so I'll do that in a separate patch later (perhaps we could purge the types from the whole module in one go?) BUG= TEST=Trybots, vie_ & voe_auto_test --automated Review URL: https://webrtc-codereview.appspot.com/998007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-18Replaced the _audio parameter with a strategy.phoglund@webrtc.org
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches. In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on. BUG= TEST=vie/voe_auto_test, trybots Review URL: https://webrtc-codereview.appspot.com/1001006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26Wire up CallStats to provide modules with correct RTT.mflodman@webrtc.org
BUG=769 TEST=Manual test since there is no ViE APi to get RTT for receive channels. Review URL: https://webrtc-codereview.appspot.com/937027 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3163 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-16Adding ViE CallStats to keep track of call statistics. As a start, only rtt ↵mflodman@webrtc.org
is handled. This CL will be followed by another CL connecting the dots. BUG=769 TEST=New unittest added. Review URL: https://webrtc-codereview.appspot.com/968006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3117 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-13Enable paced sender. pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/965016 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 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@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
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