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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / ae2d2486c71b0bcd41d18e6f68ee30b05999bf7c / video_engine / vie_rtp_rtcp_impl.cc
  1. b35efcc Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  2. 65e6f91 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
  3. 06077c9 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  4. 2a5d229 WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  5. 4db69af Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  6. 6cd34e5 Updates to send side streaming mode: by mikhal@webrtc.org · 11 years ago
  7. 89c3de3 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
  8. d6739c8 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
  9. be86bb6 Revert the revert in r2988 since that wasn't the issue. by mflodman@webrtc.org · 12 years ago
  10. f5197ca Reverse Merged r2884 & r2888 from trunk. by vikasmarwaha@webrtc.org · 12 years ago
  11. a7b57da Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago