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- 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
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.