1. f83a872 Revert 4597 "Don't force key frame when decoding with errors" by henrike@webrtc.org · 11 years ago
  2. bb89390 Implement window capturer for OS X. by sergeyu@chromium.org · 11 years ago
  3. c5fc6e0 Don't force key frame when decoding with errors by mikhal@webrtc.org · 11 years ago
  4. 0f911c9 Remove template usage of typeless enum in fake_encoder. by pbos@webrtc.org · 11 years ago
  5. 206c4a5 Enabling and testing RTCP CNAME in new API. by pbos@webrtc.org · 11 years ago
  6. 55afdbe Adds two tests for verifying padding and ramp-up behavior. by stefan@webrtc.org · 11 years ago
  7. 3540c82 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  8. 5f199d9 Android audio opensles: random deadlock in stopRecording(). by braveyao@webrtc.org · 11 years ago
  9. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  10. 9e8a66c Follow-up changes to kSelectiveErrors by mikhal@webrtc.org · 11 years ago
  11. 3ded8c9 Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots. by henrike@webrtc.org · 11 years ago
  12. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  13. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  14. f96e534 Call SetExecutablePath from test_main.cc by pbos@webrtc.org · 11 years ago
  15. 7deb335 Make FrameGeneratorCapturer own frame_generator. by pbos@webrtc.org · 11 years ago
  16. eb7b0c4 Merging video_full_stack_tests and video_engine_tests. by phoglund@webrtc.org · 11 years ago
  17. 25b57c0 iOS: unbreak the build following r4546 by fischman@webrtc.org · 11 years ago
  18. 67acd69 VideoSendStream SSRC test. by pbos@webrtc.org · 11 years ago
  19. a4944f2 Lock resources in event_posix.cc. by pbos@webrtc.org · 11 years ago
  20. 96ff6ab Added missing static_cast conversion. by pbos@webrtc.org · 11 years ago
  21. 8ce445e Implementation and testing of PLI in new API. by pbos@webrtc.org · 11 years ago
  22. 49bc1b8 Fixes to padding when driven by encoder. by stefan@webrtc.org · 11 years ago
  23. 3207eaa Made all integration tests use consistent naming. by phoglund@webrtc.org · 11 years ago
  24. 662ded4 Implementing APIs to set maximum and minimum for latency. by turaj@webrtc.org · 11 years ago
  25. ece3d35 Added choice of decode error mode to loopback test. by agalusza@google.com · 11 years ago
  26. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  27. 1e817c3 Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 11 years ago
  28. e807da9 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
  29. f594a6b 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
  30. e155918 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
  31. 298bbdb Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  32. fd6d89f The video capture module for iOS. by sjlee@webrtc.org · 11 years ago
  33. e416ab2 Remove ViEBase::Init() call from VideoCall. by pbos@webrtc.org · 11 years ago
  34. c2014fd Remove VideoEngine class from new VideoEngine API. by pbos@webrtc.org · 11 years ago
  35. d171544 Disable CanTransmitExtraRtpPacketsWithoutError on Windows. by pbos@webrtc.org · 11 years ago
  36. eca72bf Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors". by marpan@webrtc.org · 11 years ago
  37. 48bcf6f Disable racy part of RunsRtpRtcpTestWithoutErrors. by pbos@webrtc.org · 11 years ago
  38. c5e70b0 Add native_handle.h to gyp. by wuchengli@chromium.org · 11 years ago
  39. 73acde2 To allow the propagation of under-run in NetEq. by minyue@webrtc.org · 11 years ago
  40. 52c5c70 Replace MapWrapper with std::map<>. by pbos@webrtc.org · 11 years ago
  41. 8c8c87f Updated WebRTC version to 3.39 by elham@webrtc.org · 11 years ago
  42. 823a888 Signal when shutting down DirectTransport. by pbos@webrtc.org · 11 years ago
  43. d893b3f Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  44. fe881f6 Run loopback tests with network thread. by pbos@webrtc.org · 11 years ago
  45. bb6151d Added Opus stereo support by minyue@webrtc.org · 11 years ago
  46. f15cc82 Fix crash in screen capturer on Mac by sergeyu@chromium.org · 11 years ago
  47. 7d82c9d Hand over loopback packets to a network thread. by pbos@webrtc.org · 11 years ago
  48. 4870c02 Don't pace out packets or generate padding when the pacer is disabled. by stefan@webrtc.org · 11 years ago
  49. 705b38d Remove include_dirs from test/test.gyp. by pbos@webrtc.org · 11 years ago
  50. 03931c6 Remove unused unreferenced code in webrtc/ by pbos@webrtc.org · 11 years ago
  51. f43029b Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 11 years ago
  52. b0af417 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  53. e915174 Allowing decoding with errors, when disabling nack. by mikhal@webrtc.org · 11 years ago
  54. a6b178f Fix duplicate code by niklas.enbom@webrtc.org · 11 years ago
  55. a4a1afa Delete Channels without ChannelManager lock. by pbos@webrtc.org · 11 years ago
  56. ea5f28b Adding call to Opus PLC by tina.legrand@webrtc.org · 11 years ago
  57. 7b0ab2a Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests. by agalusza@google.com · 11 years ago
  58. b3ada15 Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  59. d7b06ec Code formatting on files touched in r4447. by pbos@webrtc.org · 11 years ago
  60. 280c0b9 Added configuration of max delay to ACM and NetEq by pwestin@webrtc.org · 11 years ago
  61. cda8ac1 Added Decoding with errors API to video_coding.h and removed unused DecodeError enum. by agalusza@google.com · 11 years ago
  62. 3166042 Add turaj@webrtc.org to NetEq owners. by turaj@webrtc.org · 11 years ago
  63. f3bae63 Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. by phoglund@webrtc.org · 11 years ago
  64. 44634a6 Disabled SsrcPropagatesCorrectly on Linux. by phoglund@webrtc.org · 11 years ago
  65. 71ffa0c Better error treatment in NetEqImpl::InsertPacketInternal() by minyue@webrtc.org · 11 years ago
  66. 2d3071f removed NetEq::EnableDtmf() by minyue@webrtc.org · 11 years ago
  67. ea7b33e * Update libjingle to 50389769. by wu@webrtc.org · 11 years ago
  68. 5978712 Invert dependency between webrtc_utility and media_file targets to reflect reality. by fischman@webrtc.org · 11 years ago
  69. 3ddbca9 Updated WebRTC version number to 3.38 by elham@webrtc.org · 11 years ago
  70. 3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  71. 043f6a8 Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. by pbos@webrtc.org · 11 years ago
  72. 78ab511 Use RtpHeaderParser in VideoCall implementation. by pbos@webrtc.org · 11 years ago
  73. ce85109 Glue code and tests for NACK in new VideoEngine API. by pbos@webrtc.org · 11 years ago
  74. 3a74d40 Fix send times in video_full_stack. by pbos@webrtc.org · 11 years ago
  75. 8704595 Add back is.FrameProvider() call lost in r4194. by pbos@webrtc.org · 11 years ago
  76. 146fd3c Remove redundant conditions key. by andrew@webrtc.org · 11 years ago
  77. 4b8077b Add one API for implementing Initial delay. by turaj@webrtc.org · 11 years ago
  78. acb00f5 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  79. 478d711 Add some virtual and OVERRIDEs in webrtc/common_audio/ by pbos@webrtc.org · 11 years ago
  80. 24add92 Fix some chromium-style warnings in webrtc/modules/audio_processing/ by pbos@webrtc.org · 11 years ago
  81. c7df0aa Fix crash in DesktopRegion::Intersect(). by sergeyu@chromium.org · 11 years ago
  82. 7affcd2 Fix some chromium-style warnings in webrtc/system_wrappers/ by pbos@webrtc.org · 11 years ago
  83. 7b2147f 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
  84. aa79e6e Unbreak clang/android build of webrtc. by fischman@webrtc.org · 11 years ago
  85. cb9a72b Adding possibility to use encoding time when trigger underuse for frame based overuse detection. by mflodman@webrtc.org · 11 years ago
  86. 5ce8723 Merge r4374 from stable to trunk. by xians@webrtc.org · 11 years ago
  87. 0e6fa8c Merge r4394 from stable to trunk. by xians@webrtc.org · 11 years ago
  88. 44f1239 Merge r4326 from stable to trunk. by xians@webrtc.org · 11 years ago
  89. 60bf21e Handel zero correlation if at the same time distortion is also zero. by turaj@webrtc.org · 11 years ago
  90. dd1b19d Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/ by pbos@webrtc.org · 11 years ago
  91. 10b3664 Fix some chromium-style warnings in webrtc/modules/desktop_capture/ by pbos@webrtc.org · 11 years ago
  92. 54042b9 Fix some chromium-style warnings in webrtc/modules/pacing/ by pbos@webrtc.org · 11 years ago
  93. 9d71e28 Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  94. 988a5b3 Fix some chromium-style warnings in webrtc/modules/remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
  95. 51cd3c7 Fix some chromium-style warnings in webrtc/modules/bitrate_controller/ by pbos@webrtc.org · 11 years ago
  96. 87ae00a Added libjingle_peerconnection_java_unittest to buildbot_tests.py by phoglund@webrtc.org · 11 years ago
  97. 3ed68d4 Move internal aec_core defines out of header. by andrew@webrtc.org · 11 years ago
  98. b3b9e5a Add svn:ignore properties for all spuriously-removed dirs on Linux64 Release (internal). by fischman@webrtc.org · 11 years ago
  99. e39e35f Correcting Turaj's email. by turaj@webrtc.org · 11 years ago
  100. 6907050 Fix some chromium-style warnings in webrtc/modules/video_coding/ by pbos@webrtc.org · 11 years ago