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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / libjingle / source / talk / refs/tags/FP2-open-16.12.0
  1. 5c7316d Merge tag android-5.1.0_r1 into AOSP_5.1_MERGE by Shreyas Narayan · 9 years ago fp2-sibon fp2-sibon-2.0.1 fp2-sibon-2.0.2 FP2-open-16.05.0 FP2-open-16.06.0 FP2-open-16.07.1 FP2-open-16.08.0 FP2-open-16.09.0 FP2-open-16.10.0 FP2-open-16.11.0 FP2-open-16.12.0 FP2-open-17.01.0 FP2-open-17.04.0
  2. da0509e Merge from Chromium at DEPS revision 267aeeb8d85c by Primiano Tucci · 10 years ago
  3. 6fd722a Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 40539b82d5a2c9bcf23d078e997ce0368160f5a3 by Android Chromium Automerger · 10 years ago
  4. 40539b8 Fix a problem in Thread::Send. by jiayl@webrtc.org · 10 years ago
  5. bcdb45b Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 47740f2c26aea1b3b7830abdcba063a12a61d009 by Android Chromium Automerger · 10 years ago
  6. 47740f2 Thread annotation of rtc::CriticalSection. by pbos@webrtc.org · 10 years ago
  7. 3ec84d3 Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
  8. 56dcc5b Change Android video renderer to maintain video aspect by glaznev@webrtc.org · 10 years ago
  9. 928c130 Switch HW video decoder to output byte buffers if video by glaznev@webrtc.org · 10 years ago
  10. a585cf0 (Auto)update libjingle 76169599-> 76176062 by buildbot@webrtc.org · 10 years ago
  11. b015440 Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
  12. 45017ae Revert "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
  13. 341ddff Remove DTMF status methods from Voice Engine by henrik.lundin@webrtc.org · 10 years ago
  14. d08c6be Skeleton for registering external encoders/decoders. by pbos@webrtc.org · 10 years ago
  15. ae69d42 Remove engine-level SetOptions. by pbos@webrtc.org · 10 years ago
  16. c72ff73 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 10 years ago
  17. 1373d9f Reapply 23529005 after fixing the build break issue (Chromium:582133002) by guoweis@webrtc.org · 10 years ago
  18. aeeb8f3 (Auto)update libjingle 75925673-> 75926712 by buildbot@webrtc.org · 10 years ago
  19. f87a435 (Auto)update libjingle 75924589-> 75925673 by buildbot@webrtc.org · 10 years ago
  20. c76cc07 (Auto)update libjingle 75922684-> 75924589 by buildbot@webrtc.org · 10 years ago
  21. 88b24bb Fix HW video decoder crash on some Android KK devices. by glaznev@webrtc.org · 10 years ago
  22. 1a6b25e 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
  23. 0d6677d Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
  24. ce45365 Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
  25. c467126 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at b25f2cd3bd9b8444d2a1d48ca26e2721b42c78e1 by Android Chromium Automerger · 10 years ago
  26. d7afd1f (Auto)update libjingle 75875619-> 75878731 by buildbot@webrtc.org · 10 years ago
  27. b25f2cd (Auto)update libjingle 75865376-> 75875619 by buildbot@webrtc.org · 10 years ago
  28. 7aa6baa (Auto)update libjingle 75854833-> 75865376 by buildbot@webrtc.org · 10 years ago
  29. 10fc3fc (Auto)update libjingle 75854418-> 75854833 by buildbot@webrtc.org · 10 years ago
  30. 4318280 (Auto)update libjingle 75852725-> 75853560 by buildbot@webrtc.org · 10 years ago
  31. 637a5ca A few fixes to avoid crash in HW codec on device orientation change. by glaznev@webrtc.org · 10 years ago
  32. 34f7659 Revert maximum video codec resolution on Android back to 720p again. by glaznev@webrtc.org · 10 years ago
  33. f6cfdbf (Auto)update libjingle 75818332-> 75837294 by buildbot@webrtc.org · 10 years ago
  34. dcbe13b Avoid writing a double/float to a string to avoid a crash. by jiayl@webrtc.org · 10 years ago
  35. 8f804c7 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
  36. a31086e Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  37. 5e89dbd Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  38. 131bfb7 Enable HW video decoding on Qualcomm devices. by glaznev@webrtc.org · 10 years ago
  39. d4644c1 talk/p2p/base: removed unused variable "port_" by henrike@webrtc.org · 10 years ago
  40. 6739a00 Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
  41. 19eb91c Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  42. 6a97b89 Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
  43. 030de95 (Auto)update libjingle 75683337-> 75695882 by buildbot@webrtc.org · 10 years ago
  44. cacae61 Java VideoRenderer class may be backed by two different native by glaznev@webrtc.org · 10 years ago
  45. 8d8b4c7 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  46. b15238e Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  47. 43d397f Recreate VideoStreams when setting resolution. by pbos@webrtc.org · 10 years ago
  48. 3d2e4a6 Add pbos@webrtc.org (myself) to talk/media/webrtc/. by pbos@webrtc.org · 10 years ago
  49. 600001c (Auto)update libjingle 75610402-> 75610402 by buildbot@webrtc.org · 10 years ago
  50. 5ff815f Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 1f59bcb2ae6b867fb2f52ff4654b137f98b30536 by Android Chromium Automerger · 10 years ago
  51. 1f59bcb Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..." by kjellander@webrtc.org · 10 years ago
  52. 109ba4e Add a target for the approved subset of rtc_base. by andrew@webrtc.org · 10 years ago
  53. 2ed33dd Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at a258a1548860c7a58007ad6a371fb05730e84c30 by Android Chromium Automerger · 10 years ago
  54. a846c20 HW video decoding optimization to better support HD resolution: by glaznev@webrtc.org · 10 years ago
  55. 6d3e4cf Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
  56. 22ea492 Make BW checks > 0 in peerconnection_unittest.cc. by pbos@webrtc.org · 10 years ago
  57. a258a15 Stop building talk/xmllite since it is no longer used. by henrike@webrtc.org · 10 years ago
  58. 117cae3 (Auto)update libjingle 75390072-> 75428737 by buildbot@webrtc.org · 10 years ago
  59. 73e98e3 Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..." by fbarchard@google.com · 10 years ago
  60. 6a9dda8 Temporary revert maximum video codec resolution back to 1080p. by glaznev@webrtc.org · 10 years ago
  61. 8bf99c4 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
  62. e77e198 (Auto)update libjingle 75302540-> 75327856 by buildbot@webrtc.org · 10 years ago
  63. ffdf3c8 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 965dd26ef1dfb1c2d5f3da91fad1c73bb73bc5fa by Android Chromium Automerger · 10 years ago
  64. e338b90 Stop building talk/sound since it is no longer used. by henrike@webrtc.org · 10 years ago
  65. 938f588 Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
  66. 965dd26 Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h." by henrikg@webrtc.org · 10 years ago
  67. 1a5e116 Revert 7145 "Stop building talk/sound since it is no longer used." by sprang@webrtc.org · 10 years ago
  68. 1c84149 Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. by henrik.lundin@webrtc.org · 10 years ago
  69. 10e2c7b Stop building talk/sound since it is no longer used. by henrike@webrtc.org · 10 years ago
  70. ffa7ab2 Fix frame rate selection for Android camera. by glaznev@webrtc.org · 10 years ago
  71. 18ce94c Put base tests in webrtc_tests.gyp by henrike@webrtc.org · 10 years ago
  72. 1b712c5 Enable shared socket for TurnPort. by jiayl@webrtc.org · 10 years ago
  73. ded08bf (Auto)update libjingle 75141932-> 75179475 by buildbot@webrtc.org · 10 years ago
  74. 966f092 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 10 years ago
  75. 83f8ee6 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
  76. 5d22694 TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. by jiayl@webrtc.org · 10 years ago
  77. 2ca5657 Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got by mallinath@webrtc.org · 10 years ago
  78. 3b7f619 Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own. by andresp@webrtc.org · 10 years ago
  79. 7140ae4 Expose VideoEncoders with webrtc/video_encoder.h. by pbos@webrtc.org · 10 years ago
  80. 87dac0a Revert 7093: "Implementing ICE Transports type handling in libjingle transport." by henrike@webrtc.org · 10 years ago
  81. c30ce01 Finish work queue in SctpDataMediaChannelTest. by pbos@webrtc.org · 10 years ago
  82. df1715c Fix a bot-breaking memory leak from early returning in ParseMediaDescription. by jiayl@webrtc.org · 10 years ago
  83. 5e09ab5 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
  84. 8232bfb (Auto)update libjingle 74955991-> 75042522 by buildbot@webrtc.org · 10 years ago
  85. 943ca45 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 13b2d035e2e7f2f18e3a4d3377bc1a09f43a4ff9 by Android Chromium Automerger · 10 years ago
  86. 265eb22 Implementing ICE Transports type handling in libjingle transport. by mallinath@webrtc.org · 10 years ago
  87. e8b9e34 Remove unnecessary include from testutils.cc. by thorcarpenter@google.com · 10 years ago
  88. 0fbcf96 (Auto)update libjingle 74873066-> 74873164 by buildbot@webrtc.org · 10 years ago
  89. 9289145 Fix webrtcvideoframe tests. by thorcarpenter@google.com · 10 years ago
  90. 6ea7f02 Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 by jiayl@webrtc.org · 10 years ago
  91. ec9284d (Auto)update libjingle 74857067-> 74860820 by buildbot@webrtc.org · 10 years ago
  92. 5bfcb05 (Auto)update libjingle 74851128-> 74857067 by buildbot@webrtc.org · 10 years ago
  93. 49a0c31 (Auto)update libjingle 74825992-> 74851128 by buildbot@webrtc.org · 10 years ago
  94. 8661d95 (Auto)update libjingle 74825084-> 74825992 by buildbot@webrtc.org · 10 years ago
  95. 0d3bdd0 Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice. by jiayl@webrtc.org · 10 years ago
  96. d4c377a Revert 7070 "TurnPort should retry allocation with a new address on error by henrike@webrtc.org · 10 years ago
  97. 469a71a Reduce maximum video resolution for Android. by glaznev@webrtc.org · 10 years ago
  98. 64453e0 TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. by jiayl@webrtc.org · 10 years ago
  99. 1e086db Fixes two issues in how we handle OfferToReceiveX for CreateOffer: by jiayl@webrtc.org · 10 years ago
  100. 32b3ea4 Abort Negotiate() if DoCreateOffer() fails. by pbos@webrtc.org · 10 years ago