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webrtc / src / 61b262c427d5747825d3582086786fab68d12a09
  1. 61b262c Disable tests according to issues: 1205,2272,2288,2290,2291 by henrike@webrtc.org · 12 years ago
  2. 7666db7 Update talk to 51242664. by henrike@webrtc.org · 12 years ago
  3. c095f51 Remove template usage of typeless enum in fake_encoder. by pbos@webrtc.org · 12 years ago
  4. 013d994 Enabling and testing RTCP CNAME in new API. by pbos@webrtc.org · 12 years ago
  5. 360e376 Adds two tests for verifying padding and ramp-up behavior. by stefan@webrtc.org · 12 years ago
  6. 3365422 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 12 years ago
  7. c028ee2 Android audio opensles: random deadlock in stopRecording(). by braveyao@webrtc.org · 12 years ago
  8. 286fe0b Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 12 years ago
  9. dbf6a81 Follow-up changes to kSelectiveErrors by mikhal@webrtc.org · 12 years ago
  10. 60bdb07 Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots. by henrike@webrtc.org · 12 years ago
  11. a0218a8 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 12 years ago
  12. cc39484 IP address display from stats. by hta@webrtc.org · 12 years ago
  13. 17018ed Added perf summary pages to the dashboard server. by phoglund@webrtc.org · 12 years ago
  14. 1a65d6c Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 12 years ago
  15. fbf0f69 Call SetExecutablePath from test_main.cc by pbos@webrtc.org · 12 years ago
  16. 4c96601 Make FrameGeneratorCapturer own frame_generator. by pbos@webrtc.org · 12 years ago
  17. abc1ed3 Merging video_full_stack_tests and video_engine_tests. by phoglund@webrtc.org · 12 years ago
  18. d0f4c21 iOS: unbreak the build following r4546 by fischman@webrtc.org · 12 years ago
  19. ebe68aa Fix memory leak in portallocatorsessionproxy_unittest. by wu@webrtc.org · 12 years ago
  20. cbdb9d1 Add comment about updating webrtc.DEPS when rolling gflags by kjellander@webrtc.org · 12 years ago
  21. 25b39ab Document updating gflags and remove code duplication. by kjellander@webrtc.org · 12 years ago
  22. 119a1cc VideoSendStream SSRC test. by pbos@webrtc.org · 12 years ago
  23. e6dc38e Lock resources in event_posix.cc. by pbos@webrtc.org · 12 years ago
  24. 62e5af4 Use a sourceforge_url for jsoncpp in DEPS. by pbos@webrtc.org · 12 years ago
  25. 7238e5f Fixes broken deps. Jsoncpp has moved from http://jsoncpp.svn.sourceforge.net to http://svn.code.sf.net by henrike@webrtc.org · 12 years ago
  26. d5f4c15 Added missing static_cast conversion. by pbos@webrtc.org · 12 years ago
  27. e7f056e Implementation and testing of PLI in new API. by pbos@webrtc.org · 12 years ago
  28. d4f607e Fixes to padding when driven by encoder. by stefan@webrtc.org · 12 years ago
  29. 32fe90b Made all integration tests use consistent naming. by phoglund@webrtc.org · 12 years ago
  30. f3bf5e0 Add suppressions file for TSan v2 by kjellander@webrtc.org · 12 years ago
  31. f1efc57 Implementing APIs to set maximum and minimum for latency. by turaj@webrtc.org · 12 years ago
  32. b655985 Added choice of decode error mode to loopback test. by agalusza@google.com · 12 years ago
  33. 28ff3ee Fix invalid cricket::SrtpStat::FailureKey::operator<() implementation. by fischman@webrtc.org · 12 years ago
  34. 166991f Suppress tsan errors on libjingle_peerconnection_unittest. by wu@webrtc.org · 12 years ago
  35. a2e0901 Suppress tsan errors. by wu@webrtc.org · 12 years ago
  36. 4d3e8b8 Update srtp error value in channel unittests. by mallinath@webrtc.org · 12 years ago
  37. 822fbd8 Update talk to 50918584. by wu@webrtc.org · 12 years ago
  38. dde7d4c Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 12 years ago
  39. 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 · 12 years ago
  40. 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 · 12 years ago
  41. 4298f73 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 12 years ago
  42. d7a4d23 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 12 years ago
  43. d690eab The video capture module for iOS. by sjlee@webrtc.org · 12 years ago
  44. 3d0019f Remove ViEBase::Init() call from VideoCall. by pbos@webrtc.org · 12 years ago
  45. fd39e13 Remove VideoEngine class from new VideoEngine API. by pbos@webrtc.org · 12 years ago
  46. d659143 Disable CanTransmitExtraRtpPacketsWithoutError on Windows. by pbos@webrtc.org · 12 years ago
  47. 62ecc20 Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors". by marpan@webrtc.org · 12 years ago
  48. 83ffb0d Added functionality in apprtc demo to close the capture device on hangup. by vikasmarwaha@webrtc.org · 12 years ago
  49. a05653b Disable racy part of RunsRtpRtcpTestWithoutErrors. by pbos@webrtc.org · 12 years ago
  50. e1051b0 Add native_handle.h to gyp. by wuchengli@chromium.org · 12 years ago
  51. db1cefc To allow the propagation of under-run in NetEq. by minyue@webrtc.org · 12 years ago
  52. 97d1a98 Remove suppressions for the cases that's already fixed. by wu@webrtc.org · 12 years ago
  53. 6603736 PeerConnection::RemoveStream now removes the local stream even when it's closed. Updated the unit test accordingly. by wu@webrtc.org · 12 years ago
  54. 32001ef PeerConnection shutdown-time fixes by fischman@webrtc.org · 12 years ago
  55. a550669 Update libjingle to 50733053. by mallinath@webrtc.org · 12 years ago
  56. 4ca7d3f Replace MapWrapper with std::map<>. by pbos@webrtc.org · 12 years ago
  57. dd14b2a libjingle gyp: signal errors during gyp time to avoid cryptic failures during build time. by fischman@webrtc.org · 12 years ago
  58. 1928d0e Updated WebRTC version to 3.39 by elham@webrtc.org · 12 years ago
  59. 468e19a Signal when shutting down DirectTransport. by pbos@webrtc.org · 12 years ago
  60. 0d94c2f Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 12 years ago
  61. 9668467 Run loopback tests with network thread. by pbos@webrtc.org · 12 years ago
  62. ecbe0aa Added Opus stereo support by minyue@webrtc.org · 12 years ago
  63. 91053e7 Update libjingle to 50654631. by wu@webrtc.org · 12 years ago
  64. bf853f2 Fix crash in screen capturer on Mac by sergeyu@chromium.org · 12 years ago
  65. 6cd9341 Hand over loopback packets to a network thread. by pbos@webrtc.org · 12 years ago
  66. 80865fd Don't pace out packets or generate padding when the pacer is disabled. by stefan@webrtc.org · 12 years ago
  67. 2ab209e Remove include_dirs from test/test.gyp. by pbos@webrtc.org · 12 years ago
  68. a3b7406 Remove unused unreferenced code in webrtc/ by pbos@webrtc.org · 12 years ago
  69. f4081ab Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 12 years ago
  70. a717ee9 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 12 years ago
  71. 64799da Allowing decoding with errors, when disabling nack. by mikhal@webrtc.org · 12 years ago
  72. e270331 Fix duplicate code by niklas.enbom@webrtc.org · 12 years ago
  73. 5a27e49 This CL will add support of passing all turn urls returned by the CEOD to PeerConnection object. by mallinath@webrtc.org · 12 years ago
  74. 58d76cb Delete Channels without ChannelManager lock. by pbos@webrtc.org · 12 years ago
  75. bd21fb5 Adding call to Opus PLC by tina.legrand@webrtc.org · 12 years ago
  76. d177c10 Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests. by agalusza@google.com · 12 years ago
  77. 676ff1e Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 12 years ago
  78. 825e9b0 talk/objc/README: s/libjingle/webrtc/ in repository path. by fischman@webrtc.org · 12 years ago
  79. a165d9c Code formatting on files touched in r4447. by pbos@webrtc.org · 12 years ago
  80. 401ef36 Added configuration of max delay to ACM and NetEq by pwestin@webrtc.org · 12 years ago
  81. c883fdc PeerConnection.java: enable setting trace & log levels from Java by fischman@webrtc.org · 12 years ago
  82. c4e1ab5 Added Decoding with errors API to video_coding.h and removed unused DecodeError enum. by agalusza@google.com · 12 years ago
  83. 0fc2558 Add turaj@webrtc.org to NetEq owners. by turaj@webrtc.org · 12 years ago
  84. 94aca5c Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. by phoglund@webrtc.org · 12 years ago
  85. bd69d1b Disabled SsrcPropagatesCorrectly on Linux. by phoglund@webrtc.org · 12 years ago
  86. 7bb5436 Better error treatment in NetEqImpl::InsertPacketInternal() by minyue@webrtc.org · 12 years ago
  87. 9721db7 removed NetEq::EnableDtmf() by minyue@webrtc.org · 12 years ago
  88. 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 · 12 years ago
  89. 9dba525 * Update libjingle to 50389769. by wu@webrtc.org · 12 years ago
  90. f696f25 Invert dependency between webrtc_utility and media_file targets to reflect reality. by fischman@webrtc.org · 12 years ago
  91. 9b8861c Updated WebRTC version number to 3.38 by elham@webrtc.org · 12 years ago
  92. 12dc1a3 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 12 years ago
  93. c3d93c6 talk/PRESUBMIT: Accept copyright years going back to 2004. by fischman@webrtc.org · 12 years ago
  94. ccdcbae Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. by pbos@webrtc.org · 12 years ago
  95. 4052370 Use RtpHeaderParser in VideoCall implementation. by pbos@webrtc.org · 12 years ago
  96. bbb07e6 Glue code and tests for NACK in new VideoEngine API. by pbos@webrtc.org · 12 years ago
  97. 7fb9ce0 Fix send times in video_full_stack. by pbos@webrtc.org · 12 years ago
  98. 735a7c8 Add back is.FrameProvider() call lost in r4194. by pbos@webrtc.org · 12 years ago
  99. 9434955 Disable P2PTransportChannelTest.* on memcheck and tsan bots due to issue 1972. by wu@webrtc.org · 12 years ago
  100. 2cbb429 Remove redundant conditions key. by andrew@webrtc.org · 12 years ago