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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / video_engine / vie_channel.cc
  1. d1d198b Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics(). by stefan@webrtc.org · 10 years ago
  2. 841ee42 Remove the old H264 code now that a new H.264 packetizer has been implemented. by stefan@webrtc.org · 10 years ago
  3. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
  4. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  5. 8c95e83 Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  6. f8ec08e Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  7. 2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  8. 65afbf3 Configure RTX send status on new modules. by pbos@webrtc.org · 10 years ago
  9. 88b558f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  10. 6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  11. 903e746 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
  12. 7f54561 Fix deadlock in RegisterPreDecodeImageCallback. by pbos@webrtc.org · 10 years ago
  13. d885109 Pointers were not dereferenced in GetRtpStatistics. by asapersson@webrtc.org · 10 years ago
  14. 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
  15. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  16. 9d10769 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
  17. 8edccce Cleaned up logging in video_coding. by stefan@webrtc.org · 10 years ago
  18. 9968131 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. by andresp@webrtc.org · 10 years ago
  19. 022615b Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
  20. 2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  21. 9aef225 Fix to get total number of sent and received rtcp packets. by asapersson@webrtc.org · 10 years ago
  22. 4a15560 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
  23. 800136d Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  24. d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  25. 7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  26. b4263e0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  27. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  28. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  29. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  30. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  31. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  32. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  33. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  34. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  35. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  36. ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  37. 1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  38. b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  39. ee234be Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  40. d964bf5 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
  41. 9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  42. 5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  43. 47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  44. 2e98d45 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  45. 3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  46. c4af4cf 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
  47. f4def77 Sending status fix for module. by asapersson@webrtc.org · 11 years ago
  48. 1bd9a7b Removed unused code. by asapersson@webrtc.org · 11 years ago
  49. 6646abd Video bandwidth not reported correctly by sprang@webrtc.org · 11 years ago
  50. 24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
  51. ecfef19 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  52. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  53. 63301bd Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  54. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  55. 31a8ce7 Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
  56. 9b7bdee Revert r4562 by elham@webrtc.org · 11 years ago
  57. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  58. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  59. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  60. ece3d35 Added choice of decode error mode to loopback test. by agalusza@google.com · 11 years ago
  61. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  62. d893b3f Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  63. f43029b Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 11 years ago
  64. b0af417 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  65. e915174 Allowing decoding with errors, when disabling nack. by mikhal@webrtc.org · 11 years ago
  66. ea7b33e * Update libjingle to 50389769. by wu@webrtc.org · 11 years ago
  67. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  68. b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  69. 46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  70. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  71. 0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
  72. 20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  73. b4c89a4 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
  74. 695ff2a Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
  75. 54b6ebc Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
  76. 4e5f983 Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
  77. 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  78. f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
  79. eef4fd5 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
  80. 453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
  81. 281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  82. ac6d919 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
  83. 06ad384 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 11 years ago
  84. 7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
  85. 67879bc WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
  86. 208a648 Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
  87. 065b64d Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
  88. 946d240 Adding RTX on source by mikhal@webrtc.org · 11 years ago
  89. 912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
  90. b793abe Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
  91. 2daec4c 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 · 11 years ago
  92. 78e450f Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
  93. 55e6f58 Stop and restart fix. by mflodman@webrtc.org · 11 years ago
  94. 0329e59 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 11 years ago
  95. 08998cd fixing nack list size calculation by mikhal@webrtc.org · 11 years ago
  96. 9d6fcb3 Adding a receive side API for buffering mode. by mikhal@webrtc.org · 11 years ago
  97. 0c66de6 Updates to send side streaming mode: by mikhal@webrtc.org · 11 years ago
  98. cd1ac8b Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 11 years ago
  99. 1619664 Adding a send side API for streaming by mikhal@webrtc.org · 11 years ago
  100. 7fff32c Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 11 years ago