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webrtc / src / 0be49d8d1028a2eb3aafd979091cfa0311e6c3c3 / webrtc / call / call.cc
  1. 559af38 Split CongestionController into send- and receive-side classes. by nisse · 8 years ago
  2. 9ea46b5 Ignore packets sent on old network route when receiving feedback. by Stefan Holmer · 8 years ago
  3. c69385d Add |protected_by_flexfec| flag to VideoReceiveStream::Config. by nisse · 8 years ago
  4. dea489f Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule. by tommi · 8 years ago
  5. 7987812 Count FlexFEC packets in Call UMA stats. by brandtr · 8 years ago
  6. 657bab2 Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  7. 5c29a7a Rename flexfec AddAndProcessReceivedPacket --> OnRtpPacket. by nisse · 8 years ago
  8. 38cc1d6 Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket. by nisse · 8 years ago
  9. 4709e89 Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call. by nisse · 8 years ago
  10. 6b34124 Remove unnecessary RTPHeaderParser, following https://codereview.webrtc.org/2659563002/ by solenberg · 8 years ago
  11. d44ce05 Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ ) by nisse · 8 years ago
  12. 14245cc Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) by nisse · 8 years ago
  13. 6d4dd59 Always call RemoteBitrateEstimator::IncomingPacket from Call. by nisse · 8 years ago
  14. 3ebbcb5 Stop using VoEVideoSync in Call/VideoReceiveStream. by solenberg · 8 years ago
  15. fb45c6c Inform jitter buffer about FlexFEC protection. by brandtr · 8 years ago
  16. 5a2c506 Set the start bitrate to the delay-based BWE. by stefan · 8 years ago
  17. 1474212 Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ ) by brandtr · 8 years ago
  18. e497495 Revert of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #7 id:160001 of https://codereview.webrtc.org/2646073004/ ) by kjellander · 8 years ago
  19. fe2bef3 Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. by brandtr · 8 years ago
  20. e256bc5 Delete left-over using declaration. by nisse · 8 years ago
  21. b935984 Revert of Move congestion controller processing to the pacer thread. (patchset #5 id:80001 of https://codereview.webrtc.org/2637783003/ ) by nisse · 8 years ago
  22. b3dc2b7 Move congestion controller processing to the pacer thread. by nisse · 8 years ago
  23. fa5a368 Let FlexfecReceiveStreamImpl send RTCP RRs. by brandtr · 8 years ago
  24. b29e652 Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )" by brandtr · 8 years ago
  25. 70e4053 Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ ) by brandtr · 8 years ago
  26. ab2ffa3 Parse FlexFEC RTP headers in Call and add integration with BWE. by brandtr · 8 years ago
  27. 7250b39 Move FlexfecReceiveStream from api/call/ to call/. by brandtr · 8 years ago
  28. 1cfbd60 Generalize FlexfecReceiveStream::Config. by brandtr · 8 years ago
  29. 446fcb6 Clean up FlexfecReceiveStream ctor signatures. by brandtr · 8 years ago
  30. f515ab8 Moved call.h and most of api/call/* into call/ by ossu · 8 years ago
  31. 9332b7d Reland "Update rtt on audio only calls". by michaelt · 8 years ago
  32. 076c011 Change unit of logged bitrate stats in bytes/s to bits/s. by asapersson · 8 years ago
  33. 78b4d56 Relanding "Pass time constant to bwe smoothing filter." by minyue · 8 years ago
  34. 0245da0 Move ownership of PacketRouter from CongestionController to Call. by nisse · 8 years ago
  35. 6287e82 Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ ) by ossu · 8 years ago
  36. 9abbf5a Pass time constanct to bwe smoothing filter. by michaelt · 8 years ago
  37. e2b1501 Start probes only after network is connected. by Sergey Ulanov · 8 years ago
  38. a814941 Fix unit for logged bitrates at the end of a call. by Åsa Persson · 8 years ago
  39. 906c5dc Revert of Start probes only after network is connected. (patchset #9 id:240001 of https://codereview.webrtc.org/2458863002/ ) by honghaiz · 8 years ago
  40. 5c99c76 Start probes only after network is connected. by sergeyu · 8 years ago
  41. 43cb716 Add ToString method to AggregatedStats and log stats at the end of a call. by asapersson · 8 years ago
  42. 7602aab Remove usage of VoEBase::AssociateSendChannel() from WVoMC. by solenberg · 8 years ago
  43. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  44. 25445d3 Integrate FlexfecReceiveStream with Call. by brandtr · 8 years ago
  45. 4e52386 Reland of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #1 id:1 of https://codereview.webrtc.org/2427733002/ ) by brandtr · 8 years ago
  46. 862d74d Revert of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #2 id:60001 of https://codereview.webrtc.org/2390823009/ ) by honghaiz · 8 years ago
  47. 9c4b4b4 Add path for recovered packets from internal::Call to RtpStreamReceiver. by brandtr · 8 years ago
  48. 982bf89 Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ ) by sprang · 8 years ago
  49. e0729c5 Add RtcpRttStats to AudioStream by michaelt · 8 years ago
  50. e0928d8 Added logging for audio send/receive stream configs. by ivoc · 8 years ago
  51. 11a9cbf Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  52. cc91d28 Moved RtcEventLog files from call/ to logging/ by skvlad · 8 years ago
  53. e035e2d Set the event log in Channel from AudioSendStream. This will re-enable logging of outgoing audio packets. by terelius · 9 years ago
  54. 52200d0 Stop increasing loss-based BWE if no feedback is received. by Stefan Holmer · 9 years ago
  55. 1d02d3e Remove RTC_LOGGED_* macro. by asapersson · 9 years ago
  56. ce2e136 Update AvgCounter to have the ability to include last period metric for subsequent intervals without samples (e.g. for non-periodic updated stats). by asapersson · 9 years ago
  57. 250fd97 Use RateCounter for received bitrate stats: by asapersson · 9 years ago
  58. 26091b1 This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads. by perkj · 9 years ago
  59. 5bed20f Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video. by Per · 9 years ago
  60. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 9 years ago
  61. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 9 years ago
  62. 2e5cfcd Add periodic logging of video stats. by asapersson · 9 years ago
  63. 4374a09 Only update codec type histogram if lifetime is long enough (10 sec). by asapersson · 9 years ago
  64. 86cc6ff Variable audio bitrate. by mflodman · 9 years ago
  65. 2638c6f Ignore zero bitrate updates in the UMA BWE stats as they represent network being down and would bias the stats. by stefan · 9 years ago
  66. 6d6122b Avoid race in Call destructor by sprang · 9 years ago
  67. be40296 Fix bug where a connection switch causes BWE to be set to zero. by Stefan Holmer · 9 years ago
  68. 9c0b551 Fix bug where min transmit bitrate wasn't working by sprang · 9 years ago
  69. 14d5dbe Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 9 years ago
  70. 9e03c3b Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 9 years ago
  71. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  72. 059e183 Reland of "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #1 id:1 of https://codereview.webrtc.org/2098703004/ ) by honghaiz · 9 years ago
  73. ae4d0d9 Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #5 id:120001 of https://codereview.webrtc.org/2041593002/ ) by honghaiz · 9 years ago
  74. 5b5d2cd Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )" by Honghai Zhang · 9 years ago
  75. 71ee44c This cl: by perkj · 9 years ago
  76. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 9 years ago
  77. 101f250 Implementing auto pausing of video streams. by mflodman · 9 years ago
  78. 72d41aa Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ ) by guidou · 9 years ago
  79. 2221e1c Update the BWE when the network route changes. by honghaiz · 9 years ago
  80. adafe0b Properly wire up the event log to the VideoSendStream. by terelius · 9 years ago
  81. ec81bcd Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate by perkj · 9 years ago
  82. e30c272 Revert "Reland of Remove SendPacer from ViEEncoder by perkj · 9 years ago
  83. 28a4456 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )" by Per · 9 years ago
  84. 825eb58 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ ) by perkj · 9 years ago
  85. 857c5cc Remove SendPacer from ViEEncoder by perkj · 9 years ago
  86. 35151f3 Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket. by asapersson · 9 years ago
  87. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  88. 1c7fdd8 Remove calls to ScopedToUnique and UniqueToScoped by kwiberg · 9 years ago
  89. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  90. 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
  91. 58d992e Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*). by asapersson · 9 years ago
  92. 7a43d25 Make the audio channel communicate network state changes to the call. by skvlad · 9 years ago
  93. b25345e Replace scoped_ptr with unique_ptr in webrtc/call/ by kwiberg · 9 years ago
  94. 86aabb2 Move BitrateAllocator reference from ViEEncoder to VideoSendStream. by mflodman · 9 years ago
  95. c379fcb Break out pacer thread from CongestionController to increase testability. by Stefan Holmer · 9 years ago
  96. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  97. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
  98. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  99. 58c664c Clean up of CongestionController. by Stefan Holmer · 9 years ago
  100. 28ba927 Switch to use new implementation in metrics.h. by asapersson · 9 years ago