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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 65bf249ea9175b65e83705a665a4acee5bf87f00 / video_engine / vie_channel.h
  1. 2fa9f7e Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 11 years ago
  2. ddbd31e Remove ViE external encryption API. by solenberg@webrtc.org · 11 years ago
  3. a68379b Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 11 years ago
  4. 224933c Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  5. ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  6. acc2e43 Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  7. 39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  8. 0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  9. ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  10. 0e4512b Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  11. 3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  12. 9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  13. cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  14. 66e84b0 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  15. f1d22d4 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  16. cf5c552 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  17. 8db148e Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  18. 8911937 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  19. 9435a17 Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  20. da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  21. 4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  22. 3fe2e7f Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  23. c2162d1 Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
  24. 7af2f81 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  25. 3ba57eb Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  26. 4a4d15b Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
  27. a3eb5f7 Revert r4562 by elham@webrtc.org · 11 years ago
  28. 6cb612c Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  29. bdc40d4 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  30. 7758945 Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  31. 96c5642 Added choice of decode error mode to loopback test. by agalusza@google.com · 11 years ago
  32. 55055d2 Update talk to 50918584. by wu@webrtc.org · 11 years ago
  33. b0fc85b Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  34. 8a11920 Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 11 years ago
  35. 334bf81 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  36. 1628267 Revert r4301 by tnakamura@webrtc.org · 11 years ago
  37. 35c7707 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  38. 45ab259 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  39. a430fef Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  40. 7401259 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
  41. 281399a Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  42. 8a3b04d - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
  43. 967320b Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
  44. ad2b368 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
  45. 06077c9 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  46. 2a5d229 WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  47. 51868ad Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
  48. e561f8c Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
  49. 9a7b9f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 12 years ago
  50. 66ccc6e Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 12 years ago
  51. 213217c Stop and restart fix. by mflodman@webrtc.org · 12 years ago
  52. cb139b1 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 12 years ago
  53. 4db69af Adding a receive side API for buffering mode. by mikhal@webrtc.org · 12 years ago
  54. 89c3de3 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 12 years ago
  55. d6739c8 Adding a send side API for streaming by mikhal@webrtc.org · 12 years ago
  56. b36efe3 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
  57. 6318790 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
  58. 32f05a7 Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
  59. be86bb6 Revert the revert in r2988 since that wasn't the issue. by mflodman@webrtc.org · 12 years ago
  60. f5197ca Reverse Merged r2884 & r2888 from trunk. by vikasmarwaha@webrtc.org · 12 years ago
  61. dc7e6cf Switching to I420VideoFrame by mikhal@webrtc.org · 12 years ago
  62. a7b57da Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago