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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / video_engine / vie_rtp_rtcp_impl.cc
  1. c928d36 Cast payload types to int for logging. by pbos@webrtc.org · 10 years ago
  2. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  3. 2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  4. 6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  5. 3468f20 Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 10 years ago
  6. 4d61a36 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 10 years ago
  7. 2f0c5f7 Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. by solenberg@webrtc.org · 10 years ago
  8. 2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  9. 3f83f9c Implement minimum transmit bitrate. by pbos@webrtc.org · 10 years ago
  10. 55a2a27 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 10 years ago
  11. 4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  12. d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  13. 7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  14. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  15. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  16. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  17. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  18. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  19. ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  20. 1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  21. cebd1d7 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
  22. b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  23. 9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  24. 5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  25. 3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  26. 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  27. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  28. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  29. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  30. f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
  31. eef4fd5 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
  32. 453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
  33. 281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  34. 7645e4d Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
  35. 0425392 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
  36. 7ab7268 Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 11 years ago
  37. d430f32 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  38. 2788107 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
  39. 7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  40. 67879bc WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  41. 9d6fcb3 Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  42. 0c66de6 Updates to send side streaming mode: by mikhal@webrtc.org · 11 years ago
  43. cd1ac8b Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
  44. 1619664 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
  45. b6d9cfc Revert the revert in r2988 since that wasn't the issue. by mflodman@webrtc.org · 12 years ago
  46. 9f269d2 Reverse Merged r2884 & r2888 from trunk. by vikasmarwaha@webrtc.org · 12 years ago
  47. b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago