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gerrit-public.fairphone.software / platform / external / webrtc / d4f607e70ad85102b77cf0beba8f11e2599e8f99
  1. d4f607e Fixes to padding when driven by encoder. by stefan@webrtc.org · 11 years ago
  2. 32fe90b Made all integration tests use consistent naming. by phoglund@webrtc.org · 11 years ago
  3. f3bf5e0 Add suppressions file for TSan v2 by kjellander@webrtc.org · 11 years ago
  4. f1efc57 Implementing APIs to set maximum and minimum for latency. by turaj@webrtc.org · 11 years ago
  5. b655985 Added choice of decode error mode to loopback test. by agalusza@google.com · 11 years ago
  6. 28ff3ee Fix invalid cricket::SrtpStat::FailureKey::operator<() implementation. by fischman@webrtc.org · 11 years ago
  7. 166991f Suppress tsan errors on libjingle_peerconnection_unittest. by wu@webrtc.org · 11 years ago
  8. a2e0901 Suppress tsan errors. by wu@webrtc.org · 11 years ago
  9. 4d3e8b8 Update srtp error value in channel unittests. by mallinath@webrtc.org · 11 years ago
  10. 822fbd8 Update talk to 50918584. by wu@webrtc.org · 11 years ago
  11. dde7d4c Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 11 years ago
  12. cc9238e Fix OSX keydown detection. I noticed that the OSX implementation differs from Linux and Windows, and it will trigger continuously for a key that is pressed down. It would totally make sense to change this to a callback driven model, but that's a bigger change. by niklas.enbom@webrtc.org · 11 years ago
  13. c927817 OpenSl bug: not matching playout and record sample rate led to high or low pitch audio (depending on if playout rate was higher or lower than record rate). by henrike@webrtc.org · 11 years ago
  14. 4298f73 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
  15. d7a4d23 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  16. d690eab The video capture module for iOS. by sjlee@webrtc.org · 11 years ago
  17. 3d0019f Remove ViEBase::Init() call from VideoCall. by pbos@webrtc.org · 11 years ago
  18. fd39e13 Remove VideoEngine class from new VideoEngine API. by pbos@webrtc.org · 11 years ago
  19. d659143 Disable CanTransmitExtraRtpPacketsWithoutError on Windows. by pbos@webrtc.org · 11 years ago
  20. 62ecc20 Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors". by marpan@webrtc.org · 11 years ago
  21. 83ffb0d Added functionality in apprtc demo to close the capture device on hangup. by vikasmarwaha@webrtc.org · 11 years ago
  22. a05653b Disable racy part of RunsRtpRtcpTestWithoutErrors. by pbos@webrtc.org · 11 years ago
  23. e1051b0 Add native_handle.h to gyp. by wuchengli@chromium.org · 11 years ago
  24. db1cefc To allow the propagation of under-run in NetEq. by minyue@webrtc.org · 11 years ago
  25. 97d1a98 Remove suppressions for the cases that's already fixed. by wu@webrtc.org · 11 years ago
  26. 6603736 PeerConnection::RemoveStream now removes the local stream even when it's closed. Updated the unit test accordingly. by wu@webrtc.org · 11 years ago
  27. 32001ef PeerConnection shutdown-time fixes by fischman@webrtc.org · 11 years ago
  28. a550669 Update libjingle to 50733053. by mallinath@webrtc.org · 11 years ago
  29. 4ca7d3f Replace MapWrapper with std::map<>. by pbos@webrtc.org · 11 years ago
  30. dd14b2a libjingle gyp: signal errors during gyp time to avoid cryptic failures during build time. by fischman@webrtc.org · 11 years ago
  31. 1928d0e Updated WebRTC version to 3.39 by elham@webrtc.org · 11 years ago
  32. 468e19a Signal when shutting down DirectTransport. by pbos@webrtc.org · 11 years ago
  33. 0d94c2f Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  34. 9668467 Run loopback tests with network thread. by pbos@webrtc.org · 11 years ago
  35. ecbe0aa Added Opus stereo support by minyue@webrtc.org · 11 years ago
  36. 91053e7 Update libjingle to 50654631. by wu@webrtc.org · 11 years ago
  37. bf853f2 Fix crash in screen capturer on Mac by sergeyu@chromium.org · 11 years ago
  38. 6cd9341 Hand over loopback packets to a network thread. by pbos@webrtc.org · 11 years ago
  39. 80865fd Don't pace out packets or generate padding when the pacer is disabled. by stefan@webrtc.org · 11 years ago
  40. 2ab209e Remove include_dirs from test/test.gyp. by pbos@webrtc.org · 11 years ago
  41. a3b7406 Remove unused unreferenced code in webrtc/ by pbos@webrtc.org · 11 years ago
  42. f4081ab Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 11 years ago
  43. a717ee9 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  44. 64799da Allowing decoding with errors, when disabling nack. by mikhal@webrtc.org · 11 years ago
  45. e270331 Fix duplicate code by niklas.enbom@webrtc.org · 11 years ago
  46. 5a27e49 This CL will add support of passing all turn urls returned by the CEOD to PeerConnection object. by mallinath@webrtc.org · 11 years ago
  47. 58d76cb Delete Channels without ChannelManager lock. by pbos@webrtc.org · 11 years ago
  48. bd21fb5 Adding call to Opus PLC by tina.legrand@webrtc.org · 11 years ago
  49. d177c10 Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests. by agalusza@google.com · 11 years ago
  50. 676ff1e Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  51. 825e9b0 talk/objc/README: s/libjingle/webrtc/ in repository path. by fischman@webrtc.org · 11 years ago
  52. a165d9c Code formatting on files touched in r4447. by pbos@webrtc.org · 11 years ago
  53. 401ef36 Added configuration of max delay to ACM and NetEq by pwestin@webrtc.org · 11 years ago
  54. c883fdc PeerConnection.java: enable setting trace & log levels from Java by fischman@webrtc.org · 11 years ago
  55. c4e1ab5 Added Decoding with errors API to video_coding.h and removed unused DecodeError enum. by agalusza@google.com · 11 years ago
  56. 0fc2558 Add turaj@webrtc.org to NetEq owners. by turaj@webrtc.org · 11 years ago
  57. 94aca5c Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. by phoglund@webrtc.org · 11 years ago
  58. bd69d1b Disabled SsrcPropagatesCorrectly on Linux. by phoglund@webrtc.org · 11 years ago
  59. 7bb5436 Better error treatment in NetEqImpl::InsertPacketInternal() by minyue@webrtc.org · 11 years ago
  60. 9721db7 removed NetEq::EnableDtmf() by minyue@webrtc.org · 11 years ago
  61. 6e7c203 Modified apprtc demo code to detect browser by checking user_agent in apprtc.py. Now we will use Mozilla stun server if FF is detected as the browser. The CL is an improvement to r4388. by vikasmarwaha@webrtc.org · 11 years ago
  62. 9dba525 * Update libjingle to 50389769. by wu@webrtc.org · 11 years ago
  63. f696f25 Invert dependency between webrtc_utility and media_file targets to reflect reality. by fischman@webrtc.org · 11 years ago
  64. 9b8861c Updated WebRTC version number to 3.38 by elham@webrtc.org · 11 years ago
  65. 12dc1a3 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  66. c3d93c6 talk/PRESUBMIT: Accept copyright years going back to 2004. by fischman@webrtc.org · 11 years ago
  67. ccdcbae Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. by pbos@webrtc.org · 11 years ago
  68. 4052370 Use RtpHeaderParser in VideoCall implementation. by pbos@webrtc.org · 11 years ago
  69. bbb07e6 Glue code and tests for NACK in new VideoEngine API. by pbos@webrtc.org · 11 years ago
  70. 7fb9ce0 Fix send times in video_full_stack. by pbos@webrtc.org · 11 years ago
  71. 735a7c8 Add back is.FrameProvider() call lost in r4194. by pbos@webrtc.org · 11 years ago
  72. 9434955 Disable P2PTransportChannelTest.* on memcheck and tsan bots due to issue 1972. by wu@webrtc.org · 11 years ago
  73. 2cbb429 Remove redundant conditions key. by andrew@webrtc.org · 11 years ago
  74. 7df9706 Add one API for implementing Initial delay. by turaj@webrtc.org · 11 years ago
  75. 89c6740 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  76. 51b2459 Add some virtual and OVERRIDEs in webrtc/common_audio/ by pbos@webrtc.org · 11 years ago
  77. 9162080 Fix some chromium-style warnings in webrtc/modules/audio_processing/ by pbos@webrtc.org · 11 years ago
  78. 4ebd8ef Supress libjingle_unittest fails on TSan. by wu@webrtc.org · 11 years ago
  79. a054569 Fix memory leak in datachannel and its test. by wu@webrtc.org · 11 years ago
  80. 0dc0f17 sscanf isn't safe with strings that aren't null-terminated. In such case, create a local copy that is null-terminated first. by wu@webrtc.org · 11 years ago
  81. 17758e9 Fix crash in DesktopRegion::Intersect(). by sergeyu@chromium.org · 11 years ago
  82. 86d7a19 ObjC PeerConnection README: note workaround needed for crbug.com/248168 by fischman@webrtc.org · 11 years ago
  83. 1bc1954 AppRTCDemo: builds using ninja on iOS for simulator and device! by fischman@webrtc.org · 11 years ago
  84. 6abb750 Delete gtest_exclude for asan which doesn't have effect with how the bots are setup now by wu@webrtc.org · 11 years ago
  85. a2a2718 Fix some chromium-style warnings in webrtc/system_wrappers/ by pbos@webrtc.org · 11 years ago
  86. a7e360e Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers). by agalusza@google.com · 11 years ago
  87. d64719d Update libjingle to 50191337. by wu@webrtc.org · 11 years ago
  88. d3ae3c7 Unbreak clang/android build of webrtc. by fischman@webrtc.org · 11 years ago
  89. 7fdbb1c We don't need to link with libssl.so when we already depend on openssl. by wu@webrtc.org · 11 years ago
  90. 27c0408 Suppressing tsan errors on libjingle_unittest and libjingle_peerconnection_unittest. by wu@webrtc.org · 11 years ago
  91. caa7024 PeerConnectionTest.java: build on android bots as well as linux ones. by fischman@webrtc.org · 11 years ago
  92. a543114 Removes no longer needed valgrind-libjingle folder. Was workaround for some bots using wrong valgrind script. by henrike@webrtc.org · 11 years ago
  93. d40b4d9 Fix libjingle memory bots by suppressing some of the errors. by wu@webrtc.org · 11 years ago
  94. d4412fe Adding possibility to use encoding time when trigger underuse for frame based overuse detection. by mflodman@webrtc.org · 11 years ago
  95. 09e8c47 Merge r4374 from stable to trunk. by xians@webrtc.org · 11 years ago
  96. 8fff1f0 Merge r4394 from stable to trunk. by xians@webrtc.org · 11 years ago
  97. 2f84afa Merge r4326 from stable to trunk. by xians@webrtc.org · 11 years ago
  98. 7126b38 Handel zero correlation if at the same time distortion is also zero. by turaj@webrtc.org · 11 years ago
  99. 2d1a55c Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/ by pbos@webrtc.org · 11 years ago
  100. e724284 Fix some chromium-style warnings in webrtc/modules/desktop_capture/ by pbos@webrtc.org · 11 years ago