1. 1795c35 Add default implementation of Add/RemoveObserver. by pbos@webrtc.org · 10 years ago
  2. 8cad943 Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" by kjellander@webrtc.org · 10 years ago
  3. 02cd306 Update isolate.gypi files + link to isolate_driver.py by kjellander@webrtc.org · 10 years ago
  4. 359d720 Allow Android apps to set video renderer scaling type. by glaznev@webrtc.org · 10 years ago
  5. 7dfb7fa Reland disallowing blocking calls on the worker thread. by jiayl@webrtc.org · 10 years ago
  6. 6266240 Disable flaky tests: by asapersson@webrtc.org · 10 years ago
  7. 34f2a9e Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 10 years ago
  8. bebc75e Fix the duplicated candidate problem when using multiple STUN servers. by jiayl@webrtc.org · 10 years ago
  9. a21d071 Reverting part of by thorcarpenter@google.com · 10 years ago
  10. 0530511 Explicitly initialize SSL for tests. by pbos@webrtc.org · 10 years ago
  11. 3987b6d Fix a problem in Thread::Send. by jiayl@webrtc.org · 10 years ago
  12. d60d79a Thread annotation of rtc::CriticalSection. by pbos@webrtc.org · 10 years ago
  13. 38344ed Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
  14. 8166fae Change Android video renderer to maintain video aspect by glaznev@webrtc.org · 10 years ago
  15. 90668b1 Switch HW video decoder to output byte buffers if video by glaznev@webrtc.org · 10 years ago
  16. 1b7dcc1 (Auto)update libjingle 76169599-> 76176062 by buildbot@webrtc.org · 10 years ago
  17. 2c1bcea Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
  18. 3987f10 Revert "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
  19. bf7b9e0 Remove DTMF status methods from Voice Engine by henrik.lundin@webrtc.org · 10 years ago
  20. 0a2087a Skeleton for registering external encoders/decoders. by pbos@webrtc.org · 10 years ago
  21. 83f95ba Remove engine-level SetOptions. by pbos@webrtc.org · 10 years ago
  22. 64a2f10 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 10 years ago
  23. 97ed393 Reapply 23529005 after fixing the build break issue (Chromium:582133002) by guoweis@webrtc.org · 10 years ago
  24. ed5ca1f (Auto)update libjingle 75925673-> 75926712 by buildbot@webrtc.org · 10 years ago
  25. c98f217 (Auto)update libjingle 75924589-> 75925673 by buildbot@webrtc.org · 10 years ago
  26. 0c9fe72 (Auto)update libjingle 75922684-> 75924589 by buildbot@webrtc.org · 10 years ago
  27. ebf2757 Fix HW video decoder crash on some Android KK devices. by glaznev@webrtc.org · 10 years ago
  28. c1eebfa Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc. by thorcarpenter@google.com · 10 years ago
  29. e658124 Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
  30. bbe0a85 Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
  31. 6e5c784 (Auto)update libjingle 75875619-> 75878731 by buildbot@webrtc.org · 10 years ago
  32. b5a5c44 (Auto)update libjingle 75865376-> 75875619 by buildbot@webrtc.org · 10 years ago
  33. d7acf11 (Auto)update libjingle 75854833-> 75865376 by buildbot@webrtc.org · 10 years ago
  34. ccb3e3f (Auto)update libjingle 75854418-> 75854833 by buildbot@webrtc.org · 10 years ago
  35. dcc1f04 (Auto)update libjingle 75852725-> 75853560 by buildbot@webrtc.org · 10 years ago
  36. 0b435ba A few fixes to avoid crash in HW codec on device orientation change. by glaznev@webrtc.org · 10 years ago
  37. 83af77b Revert maximum video codec resolution on Android back to 720p again. by glaznev@webrtc.org · 10 years ago
  38. 933d88a (Auto)update libjingle 75818332-> 75837294 by buildbot@webrtc.org · 10 years ago
  39. 42731bd Avoid writing a double/float to a string to avoid a crash. by jiayl@webrtc.org · 10 years ago
  40. 6cd6ba8 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
  41. ab071da Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  42. 369a637 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  43. 3b67f8e Enable HW video decoding on Qualcomm devices. by glaznev@webrtc.org · 10 years ago
  44. 4a5061f talk/p2p/base: removed unused variable "port_" by henrike@webrtc.org · 10 years ago
  45. a74eda1 Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
  46. 85ef770 Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  47. ab990ae Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
  48. 6a9b155 (Auto)update libjingle 75683337-> 75695882 by buildbot@webrtc.org · 10 years ago
  49. a59c501 Java VideoRenderer class may be backed by two different native by glaznev@webrtc.org · 10 years ago
  50. 40c2aa3 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  51. f8bff76 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  52. cddd17c Recreate VideoStreams when setting resolution. by pbos@webrtc.org · 10 years ago
  53. 88e85ad Add pbos@webrtc.org (myself) to talk/media/webrtc/. by pbos@webrtc.org · 10 years ago
  54. 80132e4 (Auto)update libjingle 75610402-> 75610402 by buildbot@webrtc.org · 10 years ago
  55. 595b23c Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..." by kjellander@webrtc.org · 10 years ago
  56. 6ae5a6d Add a target for the approved subset of rtc_base. by andrew@webrtc.org · 10 years ago
  57. 9967845 HW video decoding optimization to better support HD resolution: by glaznev@webrtc.org · 10 years ago
  58. cd309e3 Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
  59. 000d867 Make BW checks > 0 in peerconnection_unittest.cc. by pbos@webrtc.org · 10 years ago
  60. 7f82635 Stop building talk/xmllite since it is no longer used. by henrike@webrtc.org · 10 years ago
  61. a42a3ad (Auto)update libjingle 75390072-> 75428737 by buildbot@webrtc.org · 10 years ago
  62. 7e31197 Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..." by fbarchard@google.com · 10 years ago
  63. 192a54f Temporary revert maximum video codec resolution back to 1080p. by glaznev@webrtc.org · 10 years ago
  64. 3decd9b Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that." by henrike@webrtc.org · 10 years ago
  65. ea77334 (Auto)update libjingle 75302540-> 75327856 by buildbot@webrtc.org · 10 years ago
  66. 1d8f780 Stop building talk/sound since it is no longer used. by henrike@webrtc.org · 10 years ago
  67. 1d53f64 Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
  68. 307d3db Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h." by henrikg@webrtc.org · 10 years ago
  69. c665dcb Revert 7145 "Stop building talk/sound since it is no longer used." by sprang@webrtc.org · 10 years ago
  70. 1972ff8 Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. by henrik.lundin@webrtc.org · 10 years ago
  71. 4c87645 Stop building talk/sound since it is no longer used. by henrike@webrtc.org · 10 years ago
  72. 3472dcd Fix frame rate selection for Android camera. by glaznev@webrtc.org · 10 years ago
  73. b2efb67 Put base tests in webrtc_tests.gyp by henrike@webrtc.org · 10 years ago
  74. b6d6928 Enable shared socket for TurnPort. by jiayl@webrtc.org · 10 years ago
  75. 5d639b3 (Auto)update libjingle 75141932-> 75179475 by buildbot@webrtc.org · 10 years ago
  76. 7d4891d Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 10 years ago
  77. 54cf150 ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that. by fbarchard@google.com · 10 years ago
  78. 22406fc TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. by jiayl@webrtc.org · 10 years ago
  79. 3d81b1b Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got by mallinath@webrtc.org · 10 years ago
  80. 4d19e05 Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own. by andresp@webrtc.org · 10 years ago
  81. b420191 Expose VideoEncoders with webrtc/video_encoder.h. by pbos@webrtc.org · 10 years ago
  82. 8b0b211 Revert 7093: "Implementing ICE Transports type handling in libjingle transport." by henrike@webrtc.org · 10 years ago
  83. 7118e61 Finish work queue in SctpDataMediaChannelTest. by pbos@webrtc.org · 10 years ago
  84. 0e52772 Fix a bot-breaking memory leak from early returning in ParseMediaDescription. by jiayl@webrtc.org · 10 years ago
  85. c172320 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. by jiayl@webrtc.org · 10 years ago
  86. fd42f9d (Auto)update libjingle 74955991-> 75042522 by buildbot@webrtc.org · 10 years ago
  87. 7256d31 Implementing ICE Transports type handling in libjingle transport. by mallinath@webrtc.org · 10 years ago
  88. cc06056 Remove unnecessary include from testutils.cc. by thorcarpenter@google.com · 10 years ago
  89. 992febb (Auto)update libjingle 74873066-> 74873164 by buildbot@webrtc.org · 10 years ago
  90. a3344cf Fix webrtcvideoframe tests. by thorcarpenter@google.com · 10 years ago
  91. ddb85ab Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 by jiayl@webrtc.org · 10 years ago
  92. af5fa95 (Auto)update libjingle 74857067-> 74860820 by buildbot@webrtc.org · 10 years ago
  93. 7e3bd3d (Auto)update libjingle 74851128-> 74857067 by buildbot@webrtc.org · 10 years ago
  94. bc6fa18 (Auto)update libjingle 74825992-> 74851128 by buildbot@webrtc.org · 10 years ago
  95. 818b7b3 (Auto)update libjingle 74825084-> 74825992 by buildbot@webrtc.org · 10 years ago
  96. dfbcf81 Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice. by jiayl@webrtc.org · 10 years ago
  97. f1427c6 Revert 7070 "TurnPort should retry allocation with a new address on error by henrike@webrtc.org · 10 years ago
  98. 4b23404 Reduce maximum video resolution for Android. by glaznev@webrtc.org · 10 years ago
  99. 574f2f6 TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. by jiayl@webrtc.org · 10 years ago
  100. 52055a2 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 10 years ago