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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / ace0823ddb33a8970e285c1be1722b51d385802f / video_engine / vie_channel.cc
  1. ace0823 Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 12 years ago
  2. 213217c Stop and restart fix. by mflodman@webrtc.org · 12 years ago
  3. cb139b1 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 12 years ago
  4. 432bc1a fixing nack list size calculation by mikhal@webrtc.org · 12 years ago
  5. 4db69af Adding a receive side API for buffering mode. by mikhal@webrtc.org · 12 years ago
  6. 6cd34e5 Updates to send side streaming mode: by mikhal@webrtc.org · 12 years ago
  7. 89c3de3 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 12 years ago
  8. d6739c8 Adding a send side API for streaming by mikhal@webrtc.org · 12 years ago
  9. a7761c7 Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 12 years ago
  10. 3442158 Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 12 years ago
  11. b36efe3 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
  12. 6318790 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
  13. 32f05a7 Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
  14. b43b611 Reorganize modules/video_render. by andrew@webrtc.org · 12 years ago
  15. be86bb6 Revert the revert in r2988 since that wasn't the issue. by mflodman@webrtc.org · 12 years ago
  16. f5197ca Reverse Merged r2884 & r2888 from trunk. by vikasmarwaha@webrtc.org · 12 years ago
  17. dc7e6cf Switching to I420VideoFrame by mikhal@webrtc.org · 12 years ago
  18. a7b57da Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago