henrika@webrtc.org
19da719a5f
Resolves TSan v2 reports data races in voe_auto_test.
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--- Note that I will add more fixes to this CL ---
BUG=1590
Review URL: https://webrtc-codereview.appspot.com/1286005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3770 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-05 14:34:57 +00:00
edjee@google.com
79b0289bfc
Adds event traces and counters for WebRTC receive side.
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Review URL: https://webrtc-codereview.appspot.com/1279005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-04 19:43:34 +00:00
stefan@webrtc.org
a27107004d
Split the NACK list into multiple RTCPs if it's too big.
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TEST=rtp_rtcp_unittests
BUG=1434
Review URL: https://webrtc-codereview.appspot.com/1148006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3609 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-05 09:02:06 +00:00
stefan@webrtc.org
a678a3baee
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
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TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1044004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-21 07:42:11 +00:00
stefan@webrtc.org
20ed36dada
Break out RtpClock to system_wrappers and make it more generic.
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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
2013-01-17 14:01:20 +00:00
stefan@webrtc.org
8d0cd07d0c
Add test to verify that padding only frames are passing through the RTP module.
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Review URL: https://webrtc-codereview.appspot.com/934023
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 14:01:46 +00:00
mflodman@webrtc.org
1c61196095
Removed not used include.
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TEST=Compiles.
Review URL: https://webrtc-codereview.appspot.com/966025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3150 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-22 09:37:27 +00:00
andrew@webrtc.org
14b43beb7c
Move src/ -> webrtc/
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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
2012-10-22 18:19:23 +00:00
