1. 8d740c8 Remove bad waiting code from video decoder release function. by glaznev@webrtc.org · 10 years ago
  2. 36b9910 (Auto)update libjingle 77263371-> 77296420 by buildbot@webrtc.org · 10 years ago
  3. aa972cf Protect send_/recv_streams_ in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  4. 3acda0f Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at 1f6d92bb25f108ee5ec418811d95ca757bf228f9 by Android Chromium Automerger · 10 years ago
  5. 1f6d92b Make the media content send only if offerToReceive is false while local streams exist. by jiayl@webrtc.org · 10 years ago
  6. 83d855e Initialize sctp_paddrparams in OpenSctpSocket(). by pbos@webrtc.org · 10 years ago
  7. d7e2a0e Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder. by glaznev@webrtc.org · 10 years ago
  8. ee8ba51 Remove potential deadlock in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  9. d253970 Isolate: Remove use of --ignore_broken_items by kjellander@webrtc.org · 10 years ago
  10. 10093cd Fixing build issue with L-sdk by henrike@webrtc.org · 10 years ago
  11. 16d9b3a talk: removes empty directories base and sound. by henrike@webrtc.org · 10 years ago
  12. 69486f2 Wire up CPU adaptation in WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
  13. 82879aa Switch to SW video decoder on Android after getting 2 or more by glaznev@webrtc.org · 10 years ago
  14. 73bd001 Revert 7355 "Fix parallelization in libjingle_p2p_unittest." by henrike@webrtc.org · 10 years ago
  15. 3cd90fb Fix parallelization in libjingle_p2p_unittest. by pbos@webrtc.org · 10 years ago
  16. bee83ea Reland "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
  17. 2d66912 Merge third_party/libjingle/source/talk from https://chromium.googlesource.com/external/webrtc/trunk/talk.git at bccb37b80d4826387bff923af409c035151b7654 by Android Chromium Automerger · 10 years ago
  18. 28fd205 Revert 7338 "Fixed the android build by making the interface pur..." by xians@webrtc.org · 10 years ago
  19. 25ebcf7 Fixed the android build by making the interface pure virtual. by xians@webrtc.org · 10 years ago
  20. da0509e Merge from Chromium at DEPS revision 267aeeb8d85c by Primiano Tucci · 10 years ago lollipop-mr1-cts-release lollipop-mr1-dev lollipop-mr1-fi-release lollipop-mr1-release lollipop-mr1-wfc-release android-5.1.0_r1 android-5.1.0_r3 android-5.1.0_r4 android-5.1.0_r5 android-5.1.1_r1 android-5.1.1_r10 android-5.1.1_r12 android-5.1.1_r13 android-5.1.1_r14 android-5.1.1_r15 android-5.1.1_r16 android-5.1.1_r17 android-5.1.1_r18 android-5.1.1_r19 android-5.1.1_r2 android-5.1.1_r20 android-5.1.1_r22 android-5.1.1_r23 android-5.1.1_r24 android-5.1.1_r25 android-5.1.1_r26 android-5.1.1_r28 android-5.1.1_r29 android-5.1.1_r3 android-5.1.1_r30 android-5.1.1_r33 android-5.1.1_r34 android-5.1.1_r35 android-5.1.1_r36 android-5.1.1_r37 android-5.1.1_r38 android-5.1.1_r4 android-5.1.1_r5 android-5.1.1_r6 android-5.1.1_r7 android-5.1.1_r8 android-5.1.1_r9 android-cts-5.1_r1 android-cts-5.1_r10 android-cts-5.1_r13 android-cts-5.1_r14 android-cts-5.1_r15 android-cts-5.1_r16 android-cts-5.1_r17 android-cts-5.1_r18 android-cts-5.1_r19 android-cts-5.1_r2 android-cts-5.1_r20 android-cts-5.1_r21 android-cts-5.1_r22 android-cts-5.1_r23 android-cts-5.1_r24 android-cts-5.1_r25 android-cts-5.1_r26 android-cts-5.1_r27 android-cts-5.1_r28 android-cts-5.1_r3 android-cts-5.1_r4 android-cts-5.1_r5 android-cts-5.1_r6 android-cts-5.1_r7 android-cts-5.1_r8 android-cts-5.1_r9
  21. 04c1c1e Add default implementation of Add/RemoveObserver. by pbos@webrtc.org · 10 years ago
  22. be51d7c Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" by kjellander@webrtc.org · 10 years ago
  23. 1e3db44 Update isolate.gypi files + link to isolate_driver.py by kjellander@webrtc.org · 10 years ago
  24. c1c0fa6 Allow Android apps to set video renderer scaling type. by glaznev@webrtc.org · 10 years ago
  25. 002a4b0 Reland disallowing blocking calls on the worker thread. by jiayl@webrtc.org · 10 years ago
  26. 697892f Disable flaky tests: by asapersson@webrtc.org · 10 years ago
  27. bccb37b Initialize SSL in unittest_main.cc. by pbos@webrtc.org · 10 years ago
  28. 4888d59 Fix the duplicated candidate problem when using multiple STUN servers. by jiayl@webrtc.org · 10 years ago
  29. 96580b0 Reverting part of by thorcarpenter@google.com · 10 years ago
  30. 9fa0b75 Explicitly initialize SSL for tests. by pbos@webrtc.org · 10 years ago
  31. 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
  32. 40539b8 Fix a problem in Thread::Send. by jiayl@webrtc.org · 10 years ago
  33. 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
  34. 47740f2 Thread annotation of rtc::CriticalSection. by pbos@webrtc.org · 10 years ago
  35. 3ec84d3 Move thread_annotations.h to webrtc/base/. by pbos@webrtc.org · 10 years ago
  36. 56dcc5b Change Android video renderer to maintain video aspect by glaznev@webrtc.org · 10 years ago
  37. 928c130 Switch HW video decoder to output byte buffers if video by glaznev@webrtc.org · 10 years ago
  38. a585cf0 (Auto)update libjingle 76169599-> 76176062 by buildbot@webrtc.org · 10 years ago
  39. b015440 Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
  40. 45017ae Revert "Remove DTMF status methods from Voice Engine" r7276 by henrik.lundin@webrtc.org · 10 years ago
  41. 341ddff Remove DTMF status methods from Voice Engine by henrik.lundin@webrtc.org · 10 years ago
  42. d08c6be Skeleton for registering external encoders/decoders. by pbos@webrtc.org · 10 years ago
  43. ae69d42 Remove engine-level SetOptions. by pbos@webrtc.org · 10 years ago
  44. c72ff73 Remove Get/SetNetEQPlayoutMode APIs by henrik.lundin@webrtc.org · 10 years ago
  45. 1373d9f Reapply 23529005 after fixing the build break issue (Chromium:582133002) by guoweis@webrtc.org · 10 years ago
  46. aeeb8f3 (Auto)update libjingle 75925673-> 75926712 by buildbot@webrtc.org · 10 years ago
  47. f87a435 (Auto)update libjingle 75924589-> 75925673 by buildbot@webrtc.org · 10 years ago
  48. c76cc07 (Auto)update libjingle 75922684-> 75924589 by buildbot@webrtc.org · 10 years ago
  49. 88b24bb Fix HW video decoder crash on some Android KK devices. by glaznev@webrtc.org · 10 years ago
  50. 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
  51. 0d6677d Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
  52. ce45365 Config struct for VideoEncoder. by pbos@webrtc.org · 10 years ago
  53. 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
  54. d7afd1f (Auto)update libjingle 75875619-> 75878731 by buildbot@webrtc.org · 10 years ago
  55. b25f2cd (Auto)update libjingle 75865376-> 75875619 by buildbot@webrtc.org · 10 years ago
  56. 7aa6baa (Auto)update libjingle 75854833-> 75865376 by buildbot@webrtc.org · 10 years ago
  57. 10fc3fc (Auto)update libjingle 75854418-> 75854833 by buildbot@webrtc.org · 10 years ago
  58. 4318280 (Auto)update libjingle 75852725-> 75853560 by buildbot@webrtc.org · 10 years ago
  59. 637a5ca A few fixes to avoid crash in HW codec on device orientation change. by glaznev@webrtc.org · 10 years ago
  60. 34f7659 Revert maximum video codec resolution on Android back to 720p again. by glaznev@webrtc.org · 10 years ago
  61. f6cfdbf (Auto)update libjingle 75818332-> 75837294 by buildbot@webrtc.org · 10 years ago
  62. dcbe13b Avoid writing a double/float to a string to avoid a crash. by jiayl@webrtc.org · 10 years ago
  63. 8f804c7 Expose VP8/H264 defaults through video_encoder.h. by pbos@webrtc.org · 10 years ago
  64. a31086e Split video_render_module implementation into default and internal implementation. by andresp@webrtc.org · 10 years ago
  65. 5e89dbd Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  66. 131bfb7 Enable HW video decoding on Qualcomm devices. by glaznev@webrtc.org · 10 years ago
  67. d4644c1 talk/p2p/base: removed unused variable "port_" by henrike@webrtc.org · 10 years ago
  68. 6739a00 Split video_capture_module specific implementation (external vs internal capture) by andresp@webrtc.org · 10 years ago
  69. 19eb91c Split video engine android initialization into each internal module initialization. by andresp@webrtc.org · 10 years ago
  70. 6a97b89 Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" by pbos@webrtc.org · 10 years ago
  71. 030de95 (Auto)update libjingle 75683337-> 75695882 by buildbot@webrtc.org · 10 years ago
  72. cacae61 Java VideoRenderer class may be backed by two different native by glaznev@webrtc.org · 10 years ago
  73. 8d8b4c7 Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  74. b15238e Implemented Network::GetBestIP() selection logic as following. by guoweis@webrtc.org · 10 years ago
  75. 43d397f Recreate VideoStreams when setting resolution. by pbos@webrtc.org · 10 years ago
  76. 3d2e4a6 Add pbos@webrtc.org (myself) to talk/media/webrtc/. by pbos@webrtc.org · 10 years ago
  77. 600001c (Auto)update libjingle 75610402-> 75610402 by buildbot@webrtc.org · 10 years ago
  78. 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
  79. 1f59bcb Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..." by kjellander@webrtc.org · 10 years ago
  80. 109ba4e Add a target for the approved subset of rtc_base. by andrew@webrtc.org · 10 years ago
  81. 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
  82. a846c20 HW video decoding optimization to better support HD resolution: by glaznev@webrtc.org · 10 years ago
  83. 6d3e4cf Enable ipv6 by default for webrtc under a Finch experiment. by guoweis@webrtc.org · 10 years ago
  84. 22ea492 Make BW checks > 0 in peerconnection_unittest.cc. by pbos@webrtc.org · 10 years ago
  85. a258a15 Stop building talk/xmllite since it is no longer used. by henrike@webrtc.org · 10 years ago
  86. 117cae3 (Auto)update libjingle 75390072-> 75428737 by buildbot@webrtc.org · 10 years ago
  87. 73e98e3 Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..." by fbarchard@google.com · 10 years ago
  88. 6a9dda8 Temporary revert maximum video codec resolution back to 1080p. by glaznev@webrtc.org · 10 years ago
  89. 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
  90. e77e198 (Auto)update libjingle 75302540-> 75327856 by buildbot@webrtc.org · 10 years ago
  91. 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
  92. e338b90 Stop building talk/sound since it is no longer used. by henrike@webrtc.org · 10 years ago
  93. 938f588 Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD. by glaznev@webrtc.org · 10 years ago
  94. 965dd26 Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h." by henrikg@webrtc.org · 10 years ago
  95. 1a5e116 Revert 7145 "Stop building talk/sound since it is no longer used." by sprang@webrtc.org · 10 years ago
  96. 1c84149 Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. by henrik.lundin@webrtc.org · 10 years ago
  97. 10e2c7b Stop building talk/sound since it is no longer used. by henrike@webrtc.org · 10 years ago
  98. ffa7ab2 Fix frame rate selection for Android camera. by glaznev@webrtc.org · 10 years ago
  99. 18ce94c Put base tests in webrtc_tests.gyp by henrike@webrtc.org · 10 years ago
  100. 1b712c5 Enable shared socket for TurnPort. by jiayl@webrtc.org · 10 years ago