- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
buffered at the sender. When the buffer grows above the target delay
encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
the pacer to faster get rid of the queue after a network down event.
(Work based on issue 1237004)
BUG=1524
TESTS=trybots,vie_auto_test
Review URL: https://webrtc-codereview.appspot.com/1258004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
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
