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webrtc / src / bfd398ccda27550629ec2440888f4083e4510069 / webrtc / call
  1. 5bed20f Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video. by Per · 9 years ago
  2. 649a21a Disable RampUpTest.UpDownUpThreeStreams. by philipel · 9 years ago
  3. 8c16520 Method to parse event log directly from a string. by terelius · 9 years ago
  4. 8eb37a3 Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ ) by perkj · 9 years ago
  5. 7522a28 Removed old probe cluster logic and logic related to ssrcs from DelayBasedBwe. by philipel · 9 years ago
  6. cc16836 - Add task queue to Call with the intent of replacing the use of one of the process threads. by perkj · 9 years ago
  7. 2e5cfcd Add periodic logging of video stats. by asapersson · 9 years ago
  8. bcdad0f Generate random rtp packets with RtpPacketToSend instead of RtpSender by Danil Chapovalov · 9 years ago
  9. 76a44d5 Log current time if we stop logging before stop_time_. by terelius · 9 years ago
  10. 1318103 Reland: Add BWE plot to event log analyzer. by Stefan Holmer · 9 years ago
  11. 737336d Add NACK rate throttling for audio channels. by Erik Språng · 9 years ago
  12. ec4f068 Style cleanups in RtpSender. by Sergey Ulanov · 9 years ago
  13. 59c1ad5 Revert of Add BWE plot to event log analyzer. (patchset #6 id:100001 of https://codereview.webrtc.org/2188033004/ ) by terelius · 9 years ago
  14. 2beea2a Add BWE plot to event log analyzer. by Stefan Holmer · 9 years ago
  15. 4374a09 Only update codec type histogram if lifetime is long enough (10 sec). by asapersson · 9 years ago
  16. 86cc6ff Variable audio bitrate. by mflodman · 9 years ago
  17. 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
  18. cd349d9 Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ ) by sprang · 9 years ago
  19. 6d6122b Avoid race in Call destructor by sprang · 9 years ago
  20. a49f110 Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ ) by aluebs · 9 years ago
  21. 05ce4ae Reland Issue 2061423003: Refactor NACK bitrate allocation by Erik Språng · 9 years ago
  22. e5dd441 Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ ) by sprang · 9 years ago
  23. 5fc59e8 Refactor NACK bitrate allocation by Erik Språng · 9 years ago
  24. be40296 Fix bug where a connection switch causes BWE to be set to zero. by Stefan Holmer · 9 years ago
  25. 9b522f8 Add more logging about the bwe state and VideoSendStream state. by perkj · 9 years ago
  26. 9c0b551 Fix bug where min transmit bitrate wasn't working by sprang · 9 years ago
  27. 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
  28. 48a4beb Auto pause video streams based on encoder target bitrate. by mflodman · 9 years ago
  29. 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
  30. 1895526 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  31. 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
  32. 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
  33. 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
  34. 57c21f9 Remove ViEEncoder::Pause / Start by perkj · 9 years ago
  35. 371b43b Changes synchronization offset perfomance tracking by Danil Chapovalov · 9 years ago
  36. a6219cc FileWrapper[Impl] modifications and actually remove the "Impl" class. by tommi · 9 years ago
  37. 71ee44c This cl: by perkj · 9 years ago
  38. 0208322 GN: Add video_engine_tests by Peter Boström · 9 years ago
  39. 29b1a8d Moved creation of AudioDecoderFactory to inside PeerConnectionFactory. by ossu · 9 years ago
  40. 733b547 Movable support for VideoReceiveStream::Config and avoid copies. by Tommi · 9 years ago
  41. 101f250 Implementing auto pausing of video streams. by mflodman · 9 years ago
  42. bea8959 Hibernate the thread if there are no events in the queue. Wake it up when an event is added to the queue. by terelius · 9 years ago
  43. 14897d0 Add missing dependencies on audio, video and call to the new GN files. by katrielc · 9 years ago
  44. 46b89b9 Collapse most stdout spammy output of webrtc_perf_tests with PrintResultList by danilchap · 9 years ago
  45. 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
  46. 2221e1c Update the BWE when the network route changes. by honghaiz · 9 years ago
  47. 43587e3 Take ownership and close the platform file even if we fail to start logging. by terelius · 9 years ago
  48. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  49. adafe0b Properly wire up the event log to the VideoSendStream. by terelius · 9 years ago
  50. f1a9a54 Improved error checking for file errors in RtcEventLog. by terelius · 9 years ago
  51. 01d70a3 Add a default implementation in metrics_default.cc of histograms methods in system_wrappers/interface/metrics.h. by asapersson · 9 years ago
  52. 00b9d21 Set ViEEncoder sink_ on construction. by Peter Boström · 9 years ago
  53. fd8be34 Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  54. fea9309 This reland https://codereview.webrtc.org/1932683002/. by perkj · 9 years ago
  55. c1513ee Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API. by ivoc · 9 years ago
  56. 2ebe5b1 Refactor before implementing per stream suspension. by mflodman · 9 years ago
  57. d5c1a0b New parser for event log. Manually parse the outermost EventStream to more easily deal with corrupt or partially written logs. by terelius · 9 years ago
  58. ec81bcd Remove SendPacer from ViEEncoder and make sure SendPacer starts at a valid bitrate by perkj · 9 years ago
  59. e30c272 Revert "Reland of Remove SendPacer from ViEEncoder by perkj · 9 years ago
  60. 28a4456 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )" by Per · 9 years ago
  61. 825eb58 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ ) by perkj · 9 years ago
  62. 857c5cc Remove SendPacer from ViEEncoder by perkj · 9 years ago
  63. 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
  64. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 9 years ago
  65. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 9 years ago
  66. 1c7fdd8 Remove calls to ScopedToUnique and UniqueToScoped by kwiberg · 9 years ago
  67. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  68. 037ee92 Fix test.gyp dependency. by nisse · 9 years ago
  69. 4311ba5 Refactored CL for moving the output to a separate thread. by terelius · 9 years ago
  70. 1086ed6 Disable SwitchesToASTThenBackToTOFForVideo test completely. by deadbeef · 9 years ago
  71. 0e533ef Update the call when the network route changes by Honghai Zhang · 9 years ago
  72. 844f993 Disabling SwitchesToASTThenBackToTOFForVideo test for MSan bot. by deadbeef · 9 years ago
  73. 4aa438c Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo. by minyuel · 9 years ago
  74. 58d992e Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*). by asapersson · 9 years ago
  75. 7ade7b3 Delete class webrtc::VideoRenderer and its header file. by nisse · 9 years ago
  76. 7a43d25 Make the audio channel communicate network state changes to the call. by skvlad · 9 years ago
  77. eb83a1a This is an initial cleanup step, aiming to delete the by nisse · 9 years ago
  78. 94a23f0 Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 9 years ago
  79. 56cf60e Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 9 years ago
  80. 086f851 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 9 years ago
  81. 6021fe2 Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code. by solenberg · 9 years ago
  82. f8cdd18 Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs" by asapersson · 9 years ago
  83. b25345e Replace scoped_ptr with unique_ptr in webrtc/call/ by kwiberg · 9 years ago
  84. 86aabb2 Move BitrateAllocator reference from ViEEncoder to VideoSendStream. by mflodman · 9 years ago
  85. 905f8e7 Make ReconfigureVideoEncoder void. by Peter Boström · 9 years ago
  86. ac287ee VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock. by danilchap · 9 years ago
  87. c379fcb Break out pacer thread from CongestionController to increase testability. by Stefan Holmer · 9 years ago
  88. f4d8441 Disabled flaky tests by philipel · 9 years ago
  89. 80e1207 Move congestion controller to a separate module. by Stefan Holmer · 9 years ago
  90. e2d83d6 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt() by sprang · 9 years ago
  91. 789ba92 Simplify CongestionController. by Stefan Holmer · 9 years ago
  92. 69e59e6 [rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer by danilchap · 9 years ago
  93. 62a5ccd Update bitrate only when we have incoming packet. by Stefan Holmer · 9 years ago
  94. cde5d6b removed five redundant avsync tests to make webrtc_perf_test faster by Danil Chapovalov · 9 years ago
  95. 8c66a00 Initialize VideoSendStream members in constructor. by Peter Boström · 9 years ago
  96. 9c6a0c7 Added A/V sync tests with drifting clocks. by danilchap · 9 years ago
  97. 58c664c Clean up of CongestionController. by Stefan Holmer · 9 years ago
  98. ba4c0e4 Add send-side BWE to WebRtcVoiceEngine under a finch experiment. by stefan · 9 years ago
  99. bba9dec Use separate rtp module lists for send and receive in PacketRouter. by stefan · 9 years ago
  100. 28ba927 Switch to use new implementation in metrics.h. by asapersson · 9 years ago