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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 964d78e5c80ec8e4cbaeece5f119806b1ef9ea22 / video_engine
  1. 4590177 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  2. 2f9e587 Disable all vie_auto_tests on Linux for now (take 2) by kjellander@webrtc.org · 11 years ago
  3. 8167387 Disable all automated vie_auto_tests on Linux for now by kjellander@webrtc.org · 11 years ago
  4. b748c9d Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
  5. 7e97e4c Fix for making sure that the packet in order checks are done prior to updating the last received packet state. by stefan@webrtc.org · 11 years ago
  6. 78726d1 Updated WebRTC version to 3.46 by elham@webrtc.org · 11 years ago
  7. f4def77 Sending status fix for module. by asapersson@webrtc.org · 11 years ago
  8. 1bd9a7b Removed unused code. by asapersson@webrtc.org · 11 years ago
  9. af92d3e Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  10. a191cb0 Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  11. 6baaf30 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  12. 7773eec Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  13. 6646abd Video bandwidth not reported correctly by sprang@webrtc.org · 11 years ago
  14. f00942a Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. by fischman@webrtc.org · 11 years ago
  15. 24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
  16. 4ce7590 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
  17. ecfef19 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  18. 6036f56 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
  19. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  20. 6796d68 Updated WebRTC version to 3.45 by elham@webrtc.org · 11 years ago
  21. 4633e15 Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  22. 7c46e95 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
  23. 63301bd Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  24. c5b5ad1 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
  25. 5e74d96 Have padding decay to zero if no frames are being captured. by stefan@webrtc.org · 11 years ago
  26. 51e0101 Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  27. 44bb62a Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
  28. 93cd397 Don't pad if only one stream is sent, except if auto muted. by stefan@webrtc.org · 11 years ago
  29. 6c9c551 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
  30. a24c356 Run FullStack tests without render windows. by pbos@webrtc.org · 11 years ago
  31. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  32. e2c52d7 Move ChromaGenerator to common_video/. by pbos@webrtc.org · 11 years ago
  33. 9caedd0 Android: Fixes WebRTCDemo build (missing Java code). by henrike@webrtc.org · 11 years ago
  34. cb90617 WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties. by henrike@webrtc.org · 11 years ago
  35. eeaea08 Updated WebRTC version to 3.44 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  36. bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  37. 4b14e5a Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  38. 81cd5ca VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  39. 499392c Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
  40. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  41. a6063fd Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
  42. b5d2d16 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  43. 39079d1 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  44. c5080a9 Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  45. 362e3e5 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  46. 72790c7 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
  47. f7d5a08 Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org by elham@webrtc.org · 11 years ago
  48. a89f7e8 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
  49. 890706b Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
  50. b0382ea Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
  51. ae14504 - Reset capture deltas at resolution change. by asapersson@webrtc.org · 11 years ago
  52. 3b6d2d4 Updated WebRTC version to 3.42 by elham@webrtc.org · 11 years ago
  53. 199555c Revert test change in r4808. by stefan@webrtc.org · 11 years ago
  54. d704640 Reduce flakiness in network down test. by stefan@webrtc.org · 11 years ago
  55. 0011252 Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
  56. 28a1166 Rename EngineTest to CallTest. by pbos@webrtc.org · 11 years ago
  57. 28631e7 Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 11 years ago
  58. a89566f Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 11 years ago
  59. 93b9912 Fixes a flake in network down tests. by stefan@webrtc.org · 11 years ago
  60. 1ddd57f Break out glue for old->new Transport. by pbos@webrtc.org · 11 years ago
  61. 041d54b Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
  62. bfad17e Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  63. 990c5e3 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  64. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  65. eb2d9dd Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 11 years ago
  66. fa996f2 Split up EngineTests and RampupTests. by pbos@webrtc.org · 11 years ago
  67. 0920142 Updated WebRTC version to 3.41 by elham@webrtc.org · 11 years ago
  68. 0245bee Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 11 years ago
  69. bf6d572 Rename VideoCall to Call. by pbos@webrtc.org · 11 years ago
  70. 618a0ec ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
  71. ca20f3d Clamp camera id to legal values. by fischman@webrtc.org · 11 years ago
  72. 7dc1790 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
  73. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  74. 31a8ce7 Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
  75. 06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
  76. 0020858 Remove send and receive streams when destroyed. by pbos@webrtc.org · 11 years ago
  77. 4998966 Allow unknown flags in test_main.cc. by pbos@webrtc.org · 11 years ago
  78. c77dcb0 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. by mflodman@webrtc.org · 11 years ago
  79. 1cd055c Disable EngineTest.ReceivesPliAndRecoversWithNack. by mflodman@webrtc.org · 11 years ago
  80. 9e70940 Add FakeEncoder to VideoSendStream tests. by pbos@webrtc.org · 11 years ago
  81. 324a016 Changed method name. by mflodman@webrtc.org · 11 years ago
  82. 94ef274 Renamed method. by mflodman@webrtc.org · 11 years ago
  83. 710d2e1 Function name change. by mflodman@webrtc.org · 11 years ago
  84. a594db2 Fixing capture frame race in ViECapturer. by mflodman@webrtc.org · 11 years ago
  85. ce9de71 Overuse detection based on capture-input jitter. by pbos@webrtc.org · 11 years ago
  86. 8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
  87. 9b7bdee Revert r4562 by elham@webrtc.org · 11 years ago
  88. 6203090 Updated WebRTC version to 3.40 by elham@webrtc.org · 11 years ago
  89. e2e033a Relanding 4597 - Don't force key frame when decoding with errors. by mikhal@webrtc.org · 11 years ago
  90. c179706 Remove newapi:: namespace for typenames without overlap. by pbos@webrtc.org · 11 years ago
  91. f83a872 Revert 4597 "Don't force key frame when decoding with errors" by henrike@webrtc.org · 11 years ago
  92. c5fc6e0 Don't force key frame when decoding with errors by mikhal@webrtc.org · 11 years ago
  93. 0f911c9 Remove template usage of typeless enum in fake_encoder. by pbos@webrtc.org · 11 years ago
  94. 206c4a5 Enabling and testing RTCP CNAME in new API. by pbos@webrtc.org · 11 years ago
  95. 55afdbe Adds two tests for verifying padding and ramp-up behavior. by stefan@webrtc.org · 11 years ago
  96. 3540c82 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  97. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  98. 3ded8c9 Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots. by henrike@webrtc.org · 11 years ago
  99. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  100. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago