1. a42684b Update makefiles after merge of Chromium at 39.0.2171.95 by Ben Murdoch · 10 years ago
  2. 8af00ea Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 291035ed1d8ec308ffbc81e9cd119e2f53f92f86 by Android Chromium Automerger · 10 years ago
  3. d54aa96 Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
  4. 81b7993 Update makefiles after merge of Chromium at fb34b348eead by Android Chromium Automerger · 10 years ago
  5. bd42003 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 70861e04b3580c1350c5479c9ee26469f38ff782 by Android Chromium Automerger · 10 years ago
  6. 0ab271b Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  7. 36e363e Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
  8. 8d6e944 Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  9. 63e5c51 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9a52f504918a75e4c864ac661bce1e934adba7b1 by Android Chromium Automerger · 10 years ago
  10. e508702 Update makefiles after merge of Chromium at 6a4d455b8650 by Android Chromium Automerger · 10 years ago
  11. 6dc729b Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such. by stefan@webrtc.org · 10 years ago
  12. 9cd0bbc Mark all virtual overrides in the hierarchies of RtpDump and by henrike@webrtc.org · 10 years ago
  13. 40d8f85 Update makefiles after merge of Chromium at b62471bd5180 by Android Chromium Automerger · 10 years ago
  14. 8f88adf Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 61576f53599cf7840d3c4ebab82802b90031adcd by Android Chromium Automerger · 10 years ago
  15. 6b0dab1 Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. by henrik.lundin@webrtc.org · 10 years ago
  16. 037d14d Update makefiles after merge of Chromium at a301aef21f9e by Android Chromium Automerger · 10 years ago
  17. 0889e2e Update makefiles after merge of Chromium at d0b993bb2548 by Android Chromium Automerger · 10 years ago
  18. 4022019 Update makefiles after merge of Chromium at facf66e09bf8 by Android Chromium Automerger · 10 years ago
  19. b7e5b27 Update makefiles after merge of Chromium at 457b0a1c9412 by Android Chromium Automerger · 10 years ago
  20. cb45b28 Update makefiles after merge of Chromium at 041843cbf814 by Android Chromium Automerger · 10 years ago
  21. f1234f3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 69370488385c14d73e6ae8a3d5001c42884f9275 by Android Chromium Automerger · 10 years ago
  22. 12cdd25 Fix RTT calculations for send-only channels. by stefan@webrtc.org · 10 years ago
  23. 66a45b1 Change return value for number of discarded packets to be int. by asapersson@webrtc.org · 10 years ago
  24. d01c491 Fix audio/video sync when FEC is enabled. by stefan@webrtc.org · 10 years ago
  25. 5191730 Partial revert of r7014 (Android APK refactor) by kjellander@webrtc.org · 10 years ago
  26. 95d2195 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at f8698ce1dacfdcf804809638483adb702760469c by Android Chromium Automerger · 10 years ago
  27. b9d6b2b Android APK tests built from a normal WebRTC checkout. by kjellander@webrtc.org · 10 years ago
  28. 0de7d38 GN: Implement video_engine, video_capture and video_render. by kjellander@webrtc.org · 10 years ago
  29. 2b1b7b7 Update makefiles after merge of Chromium at b241671f0248 by Android Chromium Automerger · 10 years ago
  30. 0e2b7ec Remove Android.mk build files. by pbos@webrtc.org · 10 years ago
  31. 7a2cfc5 Remove former team members from OWNERS and WATCHLISTS by kjellander@webrtc.org · 10 years ago
  32. e0ff458 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 67afd1fc176021f625e064f20ae747e23d87d727 by Android Chromium Automerger · 10 years ago
  33. 225eac0 Bump WebRTC version number. Starting now, we will be setting WebRTC major version numbers to align with Chrome. by tnakamura@webrtc.org · 10 years ago
  34. 68fe1fc Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c1696da9a74c7ed4ed793ce993352bd370cfc414 by Torne (Richard Coles) · 10 years ago
  35. c1696da Small refactor on ViE to remove redudant conditions and long ifdefs. by andresp@webrtc.org · 10 years ago
  36. f694796 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c2ef523233552340785557abce1129a0f61537eb by Android Chromium Automerger · 10 years ago
  37. d1d198b Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics(). by stefan@webrtc.org · 10 years ago
  38. c2ef523 Decreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4). by asapersson@webrtc.org · 10 years ago
  39. 5f19242 Update makefiles after merge of Chromium at 288938 by Android Chromium Automerger · 10 years ago
  40. 22c283b Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 10 years ago
  41. 841ee42 Remove the old H264 code now that a new H.264 packetizer has been implemented. by stefan@webrtc.org · 10 years ago
  42. 8661714 Update makefiles after merge of Chromium at 287308 by Android Chromium Automerger · 10 years ago
  43. fa50854 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4a1b3e3a69d349b0d3e91f607f24e02d8b975688 by Android Chromium Automerger · 10 years ago
  44. 31b38da Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. by minyue@webrtc.org · 10 years ago
  45. f3d2702 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 82383d9b14ff8e5fedf5a70229eb0ac6b512909a by Android Chromium Automerger · 10 years ago
  46. 6111d79 Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically. by stefan@webrtc.org · 10 years ago
  47. 15097fc Remove the VPM denoiser. by pbos@webrtc.org · 10 years ago
  48. 9fbd3ec Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 10 years ago
  49. 55b0f2e Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 10 years ago
  50. c928d36 Cast payload types to int for logging. by pbos@webrtc.org · 10 years ago
  51. 09da1a7 Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 10 years ago
  52. 477e6bc Update makefiles after merge of Chromium at 282385 by Android Chromium Automerger · 10 years ago
  53. 8c95e83 Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  54. 10b9861 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 138adbb0bcdab60afda25a8727e5a071abc4ae36 by Android Chromium Automerger · 10 years ago
  55. fedbe8b Thread annotations for vie_encoder.cc/.h by stefan@webrtc.org · 10 years ago
  56. f8ec08e Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 10 years ago
  57. 6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  58. 2fd91bd Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
  59. 2d4a80c Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
  60. 65afbf3 Configure RTX send status on new modules. by pbos@webrtc.org · 10 years ago
  61. c9995bc Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. by stefan@webrtc.org · 10 years ago
  62. c7343a3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at d13c3753199496aeddc73ec88548da73283c312f by Android Chromium Automerger · 10 years ago
  63. 07dc4be Removed old code and default implementations. by asapersson@webrtc.org · 10 years ago
  64. 65a971a Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. by andresp@webrtc.org · 10 years ago
  65. 88b558f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  66. b3f0584 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9ff0df06bd431ddbf595620f94ae515bbdcde2da by Android Chromium Automerger · 10 years ago
  67. 07737de Bump version number to 3.55 by tnakamura@webrtc.org · 10 years ago
  68. 841f8c8 Update makefiles after merge of Chromium at 279716 by Android Chromium Automerger · 10 years ago
  69. 4c21d3a Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c34b9e5d5cd44c31c4f9da649b71d0d3132cf516 by Android Chromium Automerger · 10 years ago
  70. 3610f63 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  71. 4ee6348 Add tests of texture frames in video_send_stream_test. by wuchengli@chromium.org · 10 years ago
  72. c497bcd Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 68f4c7b51ec6434b302de9e97ee01f5ccdb48aa2 by Android Chromium Automerger · 10 years ago
  73. ad3bcf4 Update makefiles after merge of Chromium at 278252 by Android Chromium Automerger · 10 years ago
  74. e5a0f26 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 847dfa535730a30d57cf26d788d31070b70a02af by Android Chromium Automerger · 10 years ago
  75. cb4fdd1 Update makefiles after merge of Chromium at 277428 by Android Chromium Automerger · 10 years ago
  76. c7fcada Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a by Android Chromium Automerger · 10 years ago
  77. eddcc63 Add max limit of number for overuses. When limit is reached always apply the rampup delay. by asapersson@webrtc.org · 10 years ago
  78. 00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  79. 19f89a1 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  80. f6eaabf Increased kMaxRampUpDelayMs (120 to 240s). by asapersson@webrtc.org · 10 years ago
  81. 6845de7 Add APIs to enable padding with redundant payloads. by stefan@webrtc.org · 10 years ago
  82. adda09e Update makefiles after merge of Chromium at 276202 by Android Chromium Automerger · 10 years ago
  83. 9cd8281 Add additional metric (relative standard deviation of encode time) for overuse detection. by asapersson@webrtc.org · 10 years ago
  84. f006e8d Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
  85. 8097a46 Update makefiles after merge of Chromium at 275833 by Android Chromium Automerger · 10 years ago
  86. 20d9f00 Update makefiles after merge of Chromium at 275661 by Android Chromium Automerger · 10 years ago
  87. daf186d ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. by fischman@webrtc.org · 10 years ago
  88. 5101f84 AppRTCDemo(android): support app (UI) & capture rotation. by fischman@webrtc.org · 10 years ago
  89. 431772f Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 81f8df9af96c6b4bf43234f2a0162146a5da6112 by Android Chromium Automerger · 10 years ago
  90. 6e6292d Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 00d9c49cb076626f711988332749a0ebe8d2a32f by Android Chromium Automerger · 10 years ago
  91. 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
  92. 00d9c49 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 10 years ago
  93. 6038f4c Update makefiles after merge of Chromium at 274467 by Android Chromium Automerger · 10 years ago
  94. bf2bd58 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at ff6b4a8eddca609ad2691b54f443b6f1e9342579 by Android Chromium Automerger · 10 years ago
  95. 52dfe97 Update makefiles after merge of Chromium at 273259 by Android Chromium Automerger · 10 years ago
  96. 47475b8 Update makefiles after merge of Chromium at 273188 by Android Chromium Automerger · 10 years ago
  97. 1bdf186 Add support of texture frames for video capturer. by wuchengli@chromium.org · 10 years ago
  98. 5424828 Revert "Add support of texture frames for video capturer." by wuchengli@chromium.org · 10 years ago
  99. a3b8c85 Add support of texture frames for video capturer. by wuchengli@chromium.org · 10 years ago
  100. 98e1ef1 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at e066d34bb747f730084f1726408ca8348ff25da7 by Android Chromium Automerger · 10 years ago