Google Git
Sign in
webrtc / src / 1df9dc39579ed736d38093c943461cd028af4f99 / talk
  1. 4177615 PeerConnection(java): replace ScopedLocalRef with ScopedLocalRefFrame and fix a local reference leak in OnMessage. by fischman@webrtc.org · 11 years ago
  2. 1794693 AppRTCDemo(android): close() the throw-away DataChannel. by fischman@webrtc.org · 11 years ago
  3. e00265e Fix a compile error on Android on sctpdataengine.cc. by wu@webrtc.org · 11 years ago
  4. f6d6ed0c Update talk to 59039880. by wu@webrtc.org · 11 years ago
  5. 000dde9 Android build: make it quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  6. af320fd The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation. by fischman@webrtc.org · 11 years ago
  7. 5b3c67e objc/README: Remove outdated advice about target_os. by fischman@webrtc.org · 11 years ago
  8. 24301a6 Update talk to 58174641 together with http://review.webrtc.org/4319005/. by wu@webrtc.org · 11 years ago
  9. 62451dc Update talk to 58157731. by mallinath@webrtc.org · 11 years ago
  10. a989080 Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  11. 2018269 Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  12. a129b6c Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  13. df7b1d6 AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  14. 9ee75e9 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). by henrike@webrtc.org · 11 years ago
  15. f41f06b PeerConnection(java): rationalize pointer-to-jlong conversion. by fischman@webrtc.org · 11 years ago
  16. 9caf276 Update talk to 58037405. by wu@webrtc.org · 11 years ago
  17. 4c3faa9 Disable a libjingle unittest which is failing after a chromium roll out. by turaj@webrtc.org · 11 years ago
  18. f9bdbe3 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  19. 77507ef Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
  20. eb7def2 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  21. 32f485b Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  22. 57a5f64 revert r5230 by sergeyu@chromium.org · 11 years ago
  23. a1b21cd Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  24. 5bc25c4 Update libjingle to 57692857 by sergeyu@chromium.org · 11 years ago
  25. e003455 RTCPeerConnection(objc): avoid leaking ICE candidate on addition. by fischman@webrtc.org · 11 years ago
  26. b43202d Disable PeerConnectionEndToEndTest for tsanv2 build. by wu@webrtc.org · 11 years ago
  27. 1977960 AppRTCDemo(ios): remove codesigning hack now that gyp signs by default. by fischman@webrtc.org · 11 years ago
  28. 364f204 Update talk to 56698267. by wu@webrtc.org · 11 years ago
  29. 183c727 Disable datachannel_unittest.cc by sergeyu@chromium.org · 11 years ago
  30. a23f0ca Update talk to 56619788 by sergeyu@chromium.org · 11 years ago
  31. 16d6254 Update talk to 56183333. by wu@webrtc.org · 11 years ago
  32. 7b273a5 PeerConnection iOS: update README instructions by fischman@webrtc.org · 11 years ago
  33. 07a6fbe Update talk to 56092586. by wu@webrtc.org · 11 years ago
  34. de30501 Update talk to 55906045. by wu@webrtc.org · 11 years ago
  35. f424cb8 Update talk to 55863981. by wu@webrtc.org · 11 years ago
  36. cecfd18 Update talk to 55821645. by wu@webrtc.org · 11 years ago
  37. 9ca93a8 Explicitly @synthesize ObjC @properties by fischman@webrtc.org · 11 years ago
  38. 850bcbe Remove frame_callback.h include in webrtcvie.h. by pbos@webrtc.org · 11 years ago
  39. 97077a3 Update libjingle to 55618622. Update libyuv to r826. by wu@webrtc.org · 11 years ago
  40. d371a29 Fix tsan failures for libjingle_unittest. by wu@webrtc.org · 11 years ago
  41. 8804a29 Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread. by wu@webrtc.org · 11 years ago
  42. 4d7116b Fix tsan failures on filevideocapturer.cc. by wu@webrtc.org · 11 years ago
  43. 31628aa Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  44. 50bc553 Reenable DTLS renegotiation unittest in libjingle. by mallinath@webrtc.org · 11 years ago
  45. 3c5d2b4 Thread::Stop() must be called before any subclass's destructor completes. by wu@webrtc.org · 11 years ago
  46. 1c82037 AppRTCDemo(android): remove vestigial mentions of PowerManager by fischman@webrtc.org · 11 years ago
  47. 1d1ffc9 Update talk to 54898858. by wu@webrtc.org · 11 years ago
  48. d1cfa71 TSan v2 suppressions and exclusions for libjingle tests. by kjellander@webrtc.org · 11 years ago
  49. 6fa456f Disabling the DTLS renegotiation test case for PeerConnection. by mallinath@webrtc.org · 11 years ago
  50. 19f27e6 Update talk to 54527154. by mallinath@webrtc.org · 11 years ago
  51. 40dfbc4 Update talk to 53984350. by wu@webrtc.org · 11 years ago
  52. 4551b79 Update libjingle to 53920541. by wu@webrtc.org · 11 years ago
  53. 7818752 Update libjingle to 53856368. by wu@webrtc.org · 11 years ago
  54. 7fca2ce Add owners to [webrtc,talk]/build and *.isolate (take 2) by kjellander@webrtc.org · 11 years ago
  55. e693818 Add isolate targets for libjingle by kjellander@webrtc.org · 11 years ago
  56. 83b9e5b Add owners to [webrtc,talk]/build and *.isolate by kjellander@webrtc.org · 11 years ago
  57. 4446134 AppRTCDemo(android): support boolean value for MediaStreamConstraints.{audio,video}. by fischman@webrtc.org · 11 years ago
  58. a7266ca Fix clang build break by fischman@webrtc.org · 11 years ago
  59. 6c82e04 Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  60. 4e65e07 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  61. ddc5a19 AppRTCDemo(android): uncaught exceptions now display a modal dialog box before killing the app. by fischman@webrtc.org · 11 years ago
  62. 7e4d0df PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 12 years ago
  63. 7e809c3 Update libjingle to CL 53496343. by mallinath@webrtc.org · 12 years ago
  64. ad81ab8 Suppress SSL error strings on mac_asan to unbreak that build by mallinath@webrtc.org · 12 years ago
  65. a27be8e Update libjingle to CL 53398036. by mallinath@webrtc.org · 12 years ago
  66. 4475905 Disable flaky RapidSpeakerChange test. by andrew@webrtc.org · 12 years ago
  67. 2f240b4 Disable some flaky libjingle base tests. by andrew@webrtc.org · 12 years ago
  68. f832a55 Disable flaky TestPartialFrameHeader. by andrew@webrtc.org · 12 years ago
  69. f0f92fa Disable flaky SendDataMultipleClocks. by andrew@webrtc.org · 12 years ago
  70. 1112c30 Update libjingle to 53057474. by mallinath@webrtc.org · 12 years ago
  71. b533a82 Disabled flaky tests. BUG=2409 R=andrew@webrtc.org, mallinath@webrtc.org by asapersson@webrtc.org · 12 years ago
  72. d29ab4e Suppress SSL error strings on mac_asan to unbreak that build by fischman@webrtc.org · 12 years ago
  73. 76fe930 Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 12 years ago
  74. ccddd0a Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 12 years ago
  75. 967bfff Update talk to 52534915. by wu@webrtc.org · 12 years ago
  76. 8d1e4d6 Increase the dtmfsender test toleration to 100ms to avoid flaky. by wu@webrtc.org · 12 years ago
  77. da79008 Disabling crashing or flaky tests in peerconnection_unittest. by stefan@webrtc.org · 12 years ago
  78. b3af8ae Verify local and remote transport description before negotiation. by mallinath@webrtc.org · 12 years ago
  79. 8a14489 Disable WebRtcSessionTest.TestCreateOfferWithSctpEnabledWithoutStreams by sergeyu@chromium.org · 12 years ago
  80. a59696b Update libjingle to 52300956 by sergeyu@chromium.org · 12 years ago
  81. 82f014a OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 12 years ago
  82. 1b476d9 Disabling channelmanager unittest. This test is causing by mallinath@webrtc.org · 12 years ago
  83. ab5a091 Fixing the build error on Windows. Problem is in coversion from size_t to int. by mallinath@webrtc.org · 12 years ago
  84. 1b15f42 Update talk to 51960985. by mallinath@webrtc.org · 12 years ago
  85. 016eec0 Unbreak build by adding new mandatory ICE username param. by fischman@webrtc.org · 12 years ago
  86. c31d4d0 AppRTCDemo(iOS): prefer ISAC as audio codec by fischman@webrtc.org · 12 years ago
  87. 9080518 Restore severity precondition to logging.h. by andrew@webrtc.org · 12 years ago
  88. ccf8b56 AppRTCDemo(android): prefer ISAC for audio codec. by fischman@webrtc.org · 12 years ago
  89. 8788167 PeerConnection Java: explicitly cast DataChannel* to jlong for Java. by fischman@webrtc.org · 12 years ago
  90. cadf904 Update talk to 51664136. by wu@webrtc.org · 12 years ago
  91. 80b56a7 Revert part of libjingle roll that caused flakiness of WebRTC tests. by sergeyu@chromium.org · 12 years ago
  92. d6fef9d Fixing SetDecodeErrorMode build error - got introduced when reverting r4562 by elham@webrtc.org · 12 years ago
  93. e3de6b1 Enable ObjC build by default and reenable 64-bit mac libjingle build by fischman@webrtc.org · 12 years ago
  94. af84d78 Initialize ssl_role_ to the default role in FakeTransportChannel by mallinath@webrtc.org · 12 years ago
  95. f1fd9d0 Fix compilation on windows after libjingle updated. by sergeyu@chromium.org · 12 years ago
  96. 492e315 Update gyp file after libjingle roll. by sergeyu@chromium.org · 12 years ago
  97. 0be6aa0 Update talk to 51314459 by sergeyu@chromium.org · 12 years ago
  98. c0b1a28 Some tests were not disabled correctly as it should be DISABLED_* not DISABLE_*. by henrike@webrtc.org · 12 years ago
  99. d26f791 AppRTCDemo(android): allow audio-only calls to test iOS interop by fischman@webrtc.org · 12 years ago
  100. 61b262c Disable tests according to issues: 1205,2272,2288,2290,2291 by henrike@webrtc.org · 12 years ago