1. a77a801 Merge from Chromium at DEPS revision 278856 by Torne (Richard Coles) · 10 years ago
  2. 99bc8f3 Merge from Chromium at DEPS revision 278205 by Torne (Richard Coles) · 10 years ago
  3. 0a0ddb3 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at e25766c6c8584153a564a6af63bc92d2adcf1e94 by Android Chromium Automerger · 10 years ago
  4. e25766c (Auto)update libjingle 69543894-> 69555283 by buildbot@webrtc.org · 10 years ago
  5. 02632fd (Auto)update libjingle 69506154-> 69515138 by buildbot@webrtc.org · 10 years ago
  6. 121ec58 Fix a memory leak in SctpDataMediaChannelTest. by jiayl@webrtc.org · 10 years ago
  7. d19ccc1 Properly shut down the SCTP stack. by jiayl@webrtc.org · 10 years ago
  8. 81f0809 Makes the sid of a closed DataChannel available to reuse per the spec. by jiayl@webrtc.org · 10 years ago
  9. 96ca749 Increasing tolerances quite a bit to fight flakes. by phoglund@webrtc.org · 10 years ago
  10. 07617d7 (Auto)update libjingle 69359922-> 69365993 by buildbot@webrtc.org · 10 years ago
  11. d25cd98 (Auto)update libjingle 69337301-> 69359922 by buildbot@webrtc.org · 10 years ago
  12. d607b29 (Auto)update libjingle 69306183-> 69323802 by buildbot@webrtc.org · 10 years ago
  13. 6de09a1 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 2e507b2e465fe7f63c74e04d136ec2e23810c61f by Android Chromium Automerger · 10 years ago
  14. f0150cc Implement RTP extension support in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  15. 0ae1298 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 75ecb2a27cbb9da53ff201c0216c91065527675e by Android Chromium Automerger · 10 years ago
  16. 0b3c33a (Auto)update libjingle 69292418-> 69293749 by buildbot@webrtc.org · 10 years ago
  17. 3d341e2 (Auto)update libjingle 69291002-> 69292418 by buildbot@webrtc.org · 10 years ago
  18. 2e507b2 (Auto)update libjingle 69278008-> 69291002 by buildbot@webrtc.org · 10 years ago
  19. a4717a4 (Auto)update libjingle 69276003-> 69278008 by buildbot@webrtc.org · 10 years ago
  20. 95c71fa (Auto)update libjingle 69260070-> 69276003 by buildbot@webrtc.org · 10 years ago
  21. 27edee5 (Auto)update libjingle 69188577-> 69260070 by buildbot@webrtc.org · 10 years ago
  22. aad3e19 Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. by glaznev@webrtc.org · 10 years ago
  23. 5a09f52 Add extra logging and latency restriction to VP8 HW encoder. by glaznev@webrtc.org · 10 years ago
  24. b449ff6 (Auto)update libjingle 69144530-> 69164179 by buildbot@webrtc.org · 10 years ago
  25. 6d33fab (Auto)update libjingle 69143161-> 69144530 by buildbot@webrtc.org · 10 years ago
  26. aa77587 Add NACK feedback parameter to WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  27. 2352cb3 Implement RTX tests+fixes in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  28. 6bb6290 Remove engine_codecs_ cache from unittests. by pbos@webrtc.org · 10 years ago
  29. 4d8e980 Fix GYP DEPTH for libjingle isolate files by kjellander@webrtc.org · 10 years ago
  30. a90dbe7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  31. 5c2c63e (Auto)update libjingle 69131548-> 69132244 by buildbot@webrtc.org · 10 years ago
  32. a43c422 Initial owners file for talk/media/webrtc/. by pbos@webrtc.org · 10 years ago
  33. ec0b1b8 (Auto)update libjingle 69102234-> 69116997 by buildbot@webrtc.org · 10 years ago
  34. 2b70e77 Revert r6420 'Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck' by jiayl@webrtc.org · 10 years ago
  35. 20576e1 Revert r6390 "Adds end to end DataChannel tests." Flaky on linux_memcheck by jiayl@webrtc.org · 10 years ago
  36. 7ac6cc6 (Auto)update libjingle 69097619-> 69099564 by buildbot@webrtc.org · 10 years ago
  37. ab9ed35 Fix the flaky RTP DataChannel test. by jiayl@webrtc.org · 10 years ago
  38. 1884535 (Auto)update libjingle 69069003-> 69082899 by buildbot@webrtc.org · 10 years ago
  39. 890e587 Fixed GetStats when local and remote track are using the same ssrc. by xians@webrtc.org · 10 years ago
  40. 75ecb2a (Auto)update libjingle 69049090-> 69054765 by buildbot@webrtc.org · 10 years ago
  41. d2ca8ce (Auto)update libjingle 69005149-> 69049090 by buildbot@webrtc.org · 10 years ago
  42. fb2b140 (Auto)update libjingle 68985065-> 69005149 by buildbot@webrtc.org · 10 years ago
  43. 7249cf2 Re-land webrtcmediaengine.cc part of r6397. by pbos@webrtc.org · 10 years ago
  44. 81fabb8 (Auto)update libjingle 68982444-> 68983526 by buildbot@webrtc.org · 10 years ago
  45. 09c019f Revert 6397 "(Auto)update libjingle 68949184-> 68982444" by minyue@webrtc.org · 10 years ago
  46. 9dab816 (Auto)update libjingle 68949184-> 68982444 by buildbot@webrtc.org · 10 years ago
  47. c52cd46 Adds end to end DataChannel tests. by jiayl@webrtc.org · 10 years ago
  48. 1c1377b Add support for NVidia VP8 HW encoder. by glaznev@webrtc.org · 10 years ago
  49. bd0d8c2 Revert 6380 "Replace libjingle_root with talk_root variable." by kjellander@webrtc.org · 10 years ago
  50. 91e3af0 (Auto)update libjingle 68891947-> 68893961 by buildbot@webrtc.org · 10 years ago
  51. 1ef658d Move WebRtcVideoEngine2 fakes to unittest header. by pbos@webrtc.org · 10 years ago
  52. b723f16 Replace libjingle_root with talk_root variable. by kjellander@webrtc.org · 10 years ago
  53. 04dadd5 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at cef4b39b7ec2035a69d46360d8101a785fd154d3 by Android Chromium Automerger · 10 years ago
  54. ae73e64 Remove unused test_env.py from isolate files + fix nss path. by kjellander@webrtc.org · 10 years ago
  55. aa7cbe2 Adds support for the "apt" format parameter and turns on the RTX feature. by stefan@webrtc.org · 10 years ago
  56. 92d181b Merge from Chromium at DEPS revision 275586 by Torne (Richard Coles) · 10 years ago
  57. 19f7dc9 Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio." by jiayl@webrtc.org · 10 years ago
  58. 596929a APPRTCDemo(objc): Remove regex parsing in favor of JSON struct. by tkchin@webrtc.org · 10 years ago
  59. 50af306 Add OpenGL Android video renderer which can display multiple by glaznev@webrtc.org · 10 years ago
  60. 444d3db Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio. by jiayl@webrtc.org · 10 years ago
  61. eb70c0e AppRTCDemo(android): support app (UI) & capture rotation. by fischman@webrtc.org · 10 years ago
  62. cef4b39 (Auto)update libjingle 68701339-> 68703656 by buildbot@webrtc.org · 10 years ago
  63. daeb064 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at b5f0570d3fef0be89c236b1a063e0da5bdfb383f by Android Chromium Automerger · 10 years ago
  64. ffd44a8 Fix C++11 -Wnarrowing in channel_unittest.cc. by pbos@webrtc.org · 10 years ago
  65. 214cc46 (Auto)update libjingle 68689052-> 68689059 by buildbot@webrtc.org · 10 years ago
  66. 7254487 Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
  67. 256edd7 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 5ffd4a6be1c7f26558ee14fcf52badce9e6d2a99 by Android Chromium Automerger · 10 years ago
  68. 57a0606 (Auto)update libjingle 68646004-> 68648993 by buildbot@webrtc.org · 10 years ago
  69. b5f0570 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
  70. 91621f8 AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct. by fischman@webrtc.org · 10 years ago
  71. b931d7e Remove static initializer from WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  72. 5ffd4a6 (Auto)update libjingle 68562943-> 68571194 by buildbot@webrtc.org · 10 years ago
  73. 5eb79e1 Fix retain cycle in RTCEAGLVideoView. by tkchin@webrtc.org · 10 years ago
  74. 8c080a4 (Auto)update libjingle 68507189-> 68543735 by buildbot@webrtc.org · 10 years ago
  75. f44966d (Auto)update libjingle 68506654-> 68507189 by buildbot@webrtc.org · 10 years ago
  76. 3904bc8 (Auto)update libjingle 68501302-> 68506654 by buildbot@webrtc.org · 10 years ago
  77. 8549b80 (Auto)update libjingle 68499439-> 68501302 by buildbot@webrtc.org · 10 years ago
  78. 7c69197 (Auto)update libjingle 68495561-> 68499439 by buildbot@webrtc.org · 10 years ago
  79. 27abd57 talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291). by henrike@webrtc.org · 10 years ago
  80. 8c1588b (Auto)update libjingle 68465410-> 68487517 by buildbot@webrtc.org · 10 years ago
  81. 24caaad PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED. by fischman@webrtc.org · 10 years ago
  82. 6caba77 Add empty webrtcmediaengine.cc. by pbos@webrtc.org · 10 years ago
  83. 695e7a4 Merge from Chromium at DEPS revision 273901 by Torne (Richard Coles) · 10 years ago
  84. 842c369 (Auto)update libjingle 68379861-> 68445177 by buildbot@webrtc.org · 10 years ago
  85. fc78ef6 Remove kMaxWaitForStatsMs from tsanv2 compilation. by pbos@webrtc.org · 10 years ago
  86. ac85501 (Auto)update libjingle 68275107-> 68379861 by buildbot@webrtc.org · 10 years ago
  87. 69adcdf Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang. by pbos@webrtc.org · 10 years ago
  88. 1ef7bf9 Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback by jiayl@webrtc.org · 10 years ago
  89. ae50b2d Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates. by jiayl@webrtc.org · 10 years ago
  90. 0baf0a7 Fix AppRTC target configuration in libjingle_examples.gyp. by tkchin@webrtc.org · 10 years ago
  91. 52348f6 Implement mac version of AppRTCDemo. by tkchin@webrtc.org · 10 years ago
  92. b278723 Fix two bugs in DataChannel state transition. by jiayl@webrtc.org · 10 years ago
  93. a5702ad (Auto)update libjingle 68230113-> 68244456 by buildbot@webrtc.org · 10 years ago
  94. 50f1b41 (Auto)update libjingle 68230011-> 68230113 by buildbot@webrtc.org · 10 years ago
  95. 5b17e9b Implement new-API test RecvStreamWithoutRtx. by pbos@webrtc.org · 10 years ago
  96. a812f30 Log default receive stream creation. by pbos@webrtc.org · 10 years ago
  97. 84f5e34 Implement and fix new-API NackIsEnabled test. by pbos@webrtc.org · 10 years ago
  98. f365c83 (Auto)update libjingle 68203780-> 68206793 by buildbot@webrtc.org · 10 years ago
  99. 742af79 Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. by jiayl@webrtc.org · 10 years ago
  100. 4cb7ef9 AppRTCDemo(android): run in full-screen & immersive mode. by fischman@webrtc.org · 10 years ago