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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 499392ce105faca0d368e8512e287e81f9392950 / video_engine
  1. 499392c Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  2. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  3. a6063fd Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  4. b5d2d16 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  5. 39079d1 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  6. c5080a9 Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  7. 362e3e5 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  8. 72790c7 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
  9. f7d5a08 Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org by elham@webrtc.org · 11 years ago
  10. a89f7e8 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
  11. 890706b Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
  12. b0382ea Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
  13. ae14504 - Reset capture deltas at resolution change. by asapersson@webrtc.org · 11 years ago
  14. 3b6d2d4 Updated WebRTC version to 3.42 by elham@webrtc.org · 11 years ago
  15. 199555c Revert test change in r4808. by stefan@webrtc.org · 11 years ago
  16. d704640 Reduce flakiness in network down test. by stefan@webrtc.org · 11 years ago
  17. 0011252 Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
  18. 28a1166 Rename EngineTest to CallTest. by pbos@webrtc.org · 11 years ago
  19. 28631e7 Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 11 years ago
  20. a89566f Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 11 years ago
  21. 93b9912 Fixes a flake in network down tests. by stefan@webrtc.org · 11 years ago
  22. 1ddd57f Break out glue for old->new Transport. by pbos@webrtc.org · 11 years ago
  23. 041d54b Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
  24. bfad17e Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  25. 990c5e3 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  26. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  27. eb2d9dd Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 11 years ago
  28. fa996f2 Split up EngineTests and RampupTests. by pbos@webrtc.org · 11 years ago
  29. 0920142 Updated WebRTC version to 3.41 by elham@webrtc.org · 11 years ago
  30. 0245bee Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 11 years ago
  31. bf6d572 Rename VideoCall to Call. by pbos@webrtc.org · 11 years ago
  32. 618a0ec ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
  33. ca20f3d Clamp camera id to legal values. by fischman@webrtc.org · 11 years ago
  34. 7dc1790 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
  35. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  36. 31a8ce7 Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
  37. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  38. 0020858 Remove send and receive streams when destroyed. by pbos@webrtc.org · 11 years ago
  39. 4998966 Allow unknown flags in test_main.cc. by pbos@webrtc.org · 11 years ago
  40. c77dcb0 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. by mflodman@webrtc.org · 11 years ago
  41. 1cd055c Disable EngineTest.ReceivesPliAndRecoversWithNack. by mflodman@webrtc.org · 11 years ago
  42. 9e70940 Add FakeEncoder to VideoSendStream tests. by pbos@webrtc.org · 11 years ago
  43. 324a016 Changed method name. by mflodman@webrtc.org · 11 years ago
  44. 94ef274 Renamed method. by mflodman@webrtc.org · 11 years ago
  45. 710d2e1 Function name change. by mflodman@webrtc.org · 11 years ago
  46. a594db2 Fixing capture frame race in ViECapturer. by mflodman@webrtc.org · 11 years ago
  47. ce9de71 Overuse detection based on capture-input jitter. by pbos@webrtc.org · 11 years ago
  48. 8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
  49. 9b7bdee Revert r4562 by elham@webrtc.org · 11 years ago
  50. 6203090 Updated WebRTC version to 3.40 by elham@webrtc.org · 11 years ago
  51. e2e033a Relanding 4597 - Don't force key frame when decoding with errors. by mikhal@webrtc.org · 11 years ago
  52. c179706 Remove newapi:: namespace for typenames without overlap. by pbos@webrtc.org · 11 years ago
  53. f83a872 Revert 4597 "Don't force key frame when decoding with errors" by henrike@webrtc.org · 11 years ago
  54. c5fc6e0 Don't force key frame when decoding with errors by mikhal@webrtc.org · 11 years ago
  55. 0f911c9 Remove template usage of typeless enum in fake_encoder. by pbos@webrtc.org · 11 years ago
  56. 206c4a5 Enabling and testing RTCP CNAME in new API. by pbos@webrtc.org · 11 years ago
  57. 55afdbe Adds two tests for verifying padding and ramp-up behavior. by stefan@webrtc.org · 11 years ago
  58. 3540c82 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  59. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  60. 3ded8c9 Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots. by henrike@webrtc.org · 11 years ago
  61. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  62. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  63. f96e534 Call SetExecutablePath from test_main.cc by pbos@webrtc.org · 11 years ago
  64. 7deb335 Make FrameGeneratorCapturer own frame_generator. by pbos@webrtc.org · 11 years ago
  65. eb7b0c4 Merging video_full_stack_tests and video_engine_tests. by phoglund@webrtc.org · 11 years ago
  66. 67acd69 VideoSendStream SSRC test. by pbos@webrtc.org · 11 years ago
  67. 96ff6ab Added missing static_cast conversion. by pbos@webrtc.org · 11 years ago
  68. 8ce445e Implementation and testing of PLI in new API. by pbos@webrtc.org · 11 years ago
  69. 3207eaa Made all integration tests use consistent naming. by phoglund@webrtc.org · 11 years ago
  70. ece3d35 Added choice of decode error mode to loopback test. by agalusza@google.com · 11 years ago
  71. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  72. 1e817c3 Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 11 years ago
  73. e155918 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
  74. 298bbdb Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  75. e416ab2 Remove ViEBase::Init() call from VideoCall. by pbos@webrtc.org · 11 years ago
  76. c2014fd Remove VideoEngine class from new VideoEngine API. by pbos@webrtc.org · 11 years ago
  77. eca72bf Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors". by marpan@webrtc.org · 11 years ago
  78. 48bcf6f Disable racy part of RunsRtpRtcpTestWithoutErrors. by pbos@webrtc.org · 11 years ago
  79. 52c5c70 Replace MapWrapper with std::map<>. by pbos@webrtc.org · 11 years ago
  80. 8c8c87f Updated WebRTC version to 3.39 by elham@webrtc.org · 11 years ago
  81. 823a888 Signal when shutting down DirectTransport. by pbos@webrtc.org · 11 years ago
  82. d893b3f Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  83. fe881f6 Run loopback tests with network thread. by pbos@webrtc.org · 11 years ago
  84. f43029b Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 11 years ago
  85. b0af417 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  86. e915174 Allowing decoding with errors, when disabling nack. by mikhal@webrtc.org · 11 years ago
  87. ea7b33e * Update libjingle to 50389769. by wu@webrtc.org · 11 years ago
  88. 3ddbca9 Updated WebRTC version number to 3.38 by elham@webrtc.org · 11 years ago
  89. 3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  90. 043f6a8 Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. by pbos@webrtc.org · 11 years ago
  91. 78ab511 Use RtpHeaderParser in VideoCall implementation. by pbos@webrtc.org · 11 years ago
  92. ce85109 Glue code and tests for NACK in new VideoEngine API. by pbos@webrtc.org · 11 years ago
  93. 3a74d40 Fix send times in video_full_stack. by pbos@webrtc.org · 11 years ago
  94. 8704595 Add back is.FrameProvider() call lost in r4194. by pbos@webrtc.org · 11 years ago
  95. acb00f5 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  96. aa79e6e Unbreak clang/android build of webrtc. by fischman@webrtc.org · 11 years ago
  97. cb9a72b Adding possibility to use encoding time when trigger underuse for frame based overuse detection. by mflodman@webrtc.org · 11 years ago
  98. 45e69ce Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
  99. d69e2f4 Access receiving_ under receive_cs critical section by braveyao@webrtc.org · 11 years ago
  100. bf76ae2 Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago