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webrtc / src / b0f4b3da055cb09813d52f417f64ce2275887fea / webrtc / video_engine / vie_rtp_rtcp_impl.cc
  1. a1bfcad Cast payload types to int for logging. by pbos@webrtc.org · 11 years ago
  2. 4ef438e Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 11 years ago
  3. 2bb1bda Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 11 years ago
  4. fbb567d Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 11 years ago
  5. 4e2806d Remove WEBRTC_TRACE uses in video_engine/ by pbos@webrtc.org · 11 years ago
  6. 24bd364 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 11 years ago
  7. 4e65602 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 · 11 years ago
  8. 3fb8f7b Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 11 years ago
  9. 3349ae0 Implement minimum transmit bitrate. by pbos@webrtc.org · 11 years ago
  10. 9fd8d87 Adds APIs for reporting pacer queuing delay. by jiayl@webrtc.org · 11 years ago
  11. 8098e07 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 11 years ago
  12. 1f64f06 Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 11 years ago
  13. 0e93257 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  14. 54ae4ff Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  15. 5ab7567 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  16. 41e2615 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  17. 341e914 Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  18. 6811b6e Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  19. 096e8d9 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  20. 2656cf9 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  21. 7f73280 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
  22. ebad765 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  23. a6ad6e5 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  24. 71f055f Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  25. 8d02f5d Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  26. dc50aae Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  27. 822fbd8 Update talk to 50918584. by wu@webrtc.org · 12 years ago
  28. aa4d96a Revert r4301 by tnakamura@webrtc.org · 12 years ago
  29. 66b2e5c Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 12 years ago
  30. a6db54d - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 12 years ago
  31. c74c3c2 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 12 years ago
  32. cb9cff0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 12 years ago
  33. f5d4cb1 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 12 years ago
  34. 4dee309 Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 12 years ago
  35. 29b2219 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 12 years ago
  36. d72262d Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 12 years ago
  37. 8ca8a71 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 12 years ago
  38. ccd4b2a Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 12 years ago
  39. 9f5ebb5 Adding a payload type for RTX. by mflodman@webrtc.org · 12 years ago
  40. b238d12 WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 12 years ago
  41. ef9f76a Adding a receive side API for buffering mode. by mikhal@webrtc.org · 12 years ago
  42. 3d305c6 Updates to send side streaming mode: by mikhal@webrtc.org · 12 years ago
  43. 4fd5527 Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 12 years ago
  44. dbe97d2 Adding a send side API for streaming by mikhal@webrtc.org · 12 years ago
  45. d6ec386 Revert the revert in r2988 since that wasn't the issue. by mflodman@webrtc.org · 12 years ago
  46. 8239ca5 Reverse Merged r2884 & r2888 from trunk. by vikasmarwaha@webrtc.org · 12 years ago
  47. 14b43be Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago[Renamed from src/video_engine/vie_rtp_rtcp_impl.cc]
  48. 15e4e34 Wire up ssrc check in ViEEncoder for intra requests. by mflodman@webrtc.org · 12 years ago
  49. 976a7e6 Adding support for jointly estimating bandwidth using all streams from the same sending client. by stefan@webrtc.org · 12 years ago
  50. 5a7507f Add API for transmission smotthening. by mflodman@webrtc.org · 13 years ago
  51. 90071dd Added API to set RTP timestamp offset extension. by mflodman@webrtc.org · 13 years ago
  52. c0496e6 Expose a function for setting bandwidth estimation parameters in ViERTP_RTCP. by astor@webrtc.org · 13 years ago
  53. f5e99db Made cpplint pass for vie_channel.* and vie_encoder.*. NOLINT is used for API changes, include guards and include files in WebRTC root. by mflodman@webrtc.org · 13 years ago
  54. 9ba151b Removed cpplint warnings from all impl-files to be able to add this check as presubmit step. I don't want to change the API right now, will come later, so there are several NOLINT comments added to get around this for now. by mflodman@webrtc.org · 13 years ago
  55. 3e820e5 Remove RTP Keep-alive from VoE and ViE. The RTP module functionality will be removed in a follow-up CL shortly. by mflodman@webrtc.org · 13 years ago
  56. 0975d21 Cleanup messy data type of unknown_payload_type by leozwang@webrtc.org · 13 years ago
  57. 439be29 Add APIs for getting receive-side estimated bandwidth and codec target rate. by stefan@webrtc.org · 13 years ago
  58. 657b2a4 Added return due to gcc complaints in r1604. by mflodman@webrtc.org · 13 years ago
  59. c80d9d9 Removed default cases causing clang errors, -Wcovered-switch-default. by mflodman@webrtc.org · 13 years ago
  60. 07b45a5 Added API for getting the send-side estimated bandwidth. by stefan@webrtc.org · 13 years ago
  61. b11424b Remove ViEShared inheritance for interface impl. by mflodman@webrtc.org · 13 years ago
  62. 6cf529d Changed REMB return value to int instead of bool. by mflodman@webrtc.org · 13 years ago
  63. 8281e7d Added RTX to ViE. Review URL: http://webrtc-codereview.appspot.com/336001 by pwestin@webrtc.org · 13 years ago
  64. 84dc3d1 Add REMB functionality to ViE. by mflodman@webrtc.org · 13 years ago
  65. a4863db Moved video_engine/main/interface to video_engine/include. by mflodman@webrtc.org · 13 years ago
  66. 8da2417 Refactored ViERenderImpl and ViERTP_RTCPImpl. by mflodman@webrtc.org · 13 years ago
  67. 471e83e Refactored ViESharedData. by mflodman@webrtc.org · 13 years ago
  68. 94ea32e Move video_engine/source* to video_engine/. No code changes except paths in gyp-files. by mflodman@webrtc.org · 13 years ago[Renamed from src/video_engine/main/source/vie_rtp_rtcp_impl.cc]
  69. fbea4e5 Solves two bandwidth estimation issues and measures the sent video bitrate. by stefan@webrtc.org · 13 years ago
  70. d0bdab0 Adding API to get sent total bitrate, FEC bitrate and NACK bitrate. by stefan@webrtc.org · 13 years ago
  71. 1da1ce0 First implementation of simulcast, adds VP8 simulcast to video engine. by pwestin@webrtc.org · 13 years ago
  72. 470e71d by niklase@google.com · 14 years ago