18a186eab2
1. Updated code to set _lastdecodedSeqNum after clean up of old frames (2/3 instances were updated, 1 was ok). 2. Updated _lastDecodedSeqNum based on empty packets that arrive after the frame which they belong to was already decoded (as was with existing code with regard to filler packets). B. Code clean up. Review URL: http://webrtc-codereview.appspot.com/78001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@237 4adac7df-926f-26a2-2b94-8c16560cd09d |
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| .. | ||
| build | ||
| common_audio | ||
| common_video | ||
| modules | ||
| system_wrappers | ||
| video_engine | ||
| voice_engine | ||
| common_settings.gypi | ||
| common_types.h | ||
| engine_configurations.h | ||
| LICENSE | ||
| LICENSE_THIRD_PARTY | ||
| PATENTS | ||
| README.chromium | ||
| typedefs.h | ||
| video_engine.gyp | ||
| voice_engine.gyp | ||
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.