168f23faa5b8a49d4dd709c6649e77d5fecf36bf - src.git

commit	168f23faa5b8a49d4dd709c6649e77d5fecf36bf	[log] [tgz]
author	stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>	Fri Jul 11 13:44:02 2014
committer	stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>	Fri Jul 11 13:44:02 2014
tree	a961a30c1a0dbaf8c84386184f8001da4adf4b68
parent	ccbed3b3c4a0f7607eadafd2c1edb7578d32f099 [diff]

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.

R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/build/tsan_suppressions.cc[diff]
webrtc/common_types.h[diff]
webrtc/config.h[diff]
webrtc/modules/pacing/include/paced_sender.h[diff]
webrtc/modules/pacing/paced_sender.cc[diff]
webrtc/modules/pacing/paced_sender_unittest.cc[diff]
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h[diff]
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender.cc[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender.h[diff]
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc[diff]
webrtc/video/end_to_end_tests.cc[diff]
webrtc/video/send_statistics_proxy.cc[diff]
webrtc/video/send_statistics_proxy.h[diff]
webrtc/video/send_statistics_proxy_unittest.cc[diff]
webrtc/video/video_send_stream.cc[diff]
webrtc/video/video_send_stream.h[diff]
webrtc/video/video_send_stream_tests.cc[diff]
webrtc/video_engine/include/vie_base.h[diff]
webrtc/video_engine/vie_base_impl.cc[diff]
webrtc/video_engine/vie_base_impl.h[diff]
webrtc/video_engine/vie_channel.cc[diff]
webrtc/video_engine/vie_channel.h[diff]
webrtc/video_send_stream.h[diff]

24 files changed

tree: a961a30c1a0dbaf8c84386184f8001da4adf4b68