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2013-11-20Deliver I420VideoFrames from VideoRender module.pbos@webrtc.org
Performance issue and simplicity, this implementation skips conversion to VideoEngine's frame format and then back again to I420VideoFrame. BUG=2526 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5140 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Rename newapi::Transport::SendRTP()->SendRtp().pbos@webrtc.org
Also fit rampup_tests.cc to use internal::TransportAdapter instead of implementing its own. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Rename RTP-extension constants.pbos@webrtc.org
BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Rename video streams' start/stop methods.pbos@webrtc.org
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}(). BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3609005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20Rename Call::Create{Receive,Send}Stream().pbos@webrtc.org
Renaming the methods to include Video. Long-term there will hopefully be AudioSendStream/AudioReceiveStreams as well. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18Rename AutoMute to SuspendBelowMinBitratehenrik.lundin@webrtc.org
Changes all instances throughout the WebRTC stack. BUG=2436 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3919004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5130 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18Hook up audio/video sync to Call.stefan@webrtc.org
Adds an end-to-end audio/video sync test. BUG=2530, 2608 TEST=trybots R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3699004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15Fix breakage after introducing new test.stefan@webrtc.org
TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3899005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5127 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15Improve Call tests for RTX.stefan@webrtc.org
Also does some refactoring to reuse RtpRtcpObserver. BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-14Implement VideoSendStream::SetCodec().pbos@webrtc.org
Removing assertion that SSRC count should be the same as the number of streams in the codec. It makes sense that you don't always use the same number of streams under one call. Dropping resolution due to CPU overuse for instance can require less streams, but the SSRCs should stay allocated so that operations can resume when not overusing any more. This change also means we can get rid of the ugly SendStreamState whose content wasn't defined. Instead we use SetCodec to change resolution etc. on the fly. Should something else have to be replaced on the fly then that functionality simply has to be implemented. BUG= R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3499005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29Make video/ only depend on video_engine_core.pbos@webrtc.org
Fixes Android/Chromium build error. Previous dependencies included VideoEngine tests that couldn't build on this configuration. BUG=2535 R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5050 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-29Stop DirectTransports in VideoSendStreamTests.pbos@webrtc.org
Prevents racy packet delivery during or after Call destruction. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3099005 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5049 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28Separate Call API/build files from video_engine/.pbos@webrtc.org
BUG=2535 R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2659004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
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