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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / b7e5b2741f9cbd68e7be54e3445d28d266477c92 / video_engine / test
  1. 0e2b7ec Remove Android.mk build files. by pbos@webrtc.org · 10 years ago
  2. 22c283b Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 10 years ago
  3. 15097fc Remove the VPM denoiser. by pbos@webrtc.org · 10 years ago
  4. 6aae61c Some refactoring inside rtp_rtcp/. by pbos@webrtc.org · 10 years ago
  5. 2d4a80c Add boilerplate code for H.264. by stefan@webrtc.org · 10 years ago
  6. 00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  7. 19f89a1 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  8. f006e8d Add kjellander@webrtc.org as OWNER for *.isolate by kjellander@webrtc.org · 10 years ago
  9. daf186d ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. by fischman@webrtc.org · 10 years ago
  10. 5101f84 AppRTCDemo(android): support app (UI) & capture rotation. by fischman@webrtc.org · 10 years ago
  11. 01d8e22 vie_autotest_android.cc: stop referring to undefined functions. by fischman@webrtc.org · 10 years ago
  12. 774b3d3 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  13. 0a9ed7c Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
  14. 28b7c07 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  15. bd49ac2 Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} by andresp@webrtc.org · 10 years ago
  16. 0b8a1c4 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  17. f33a674 Make vie/voe_auto_test accept non-supported flags without error. by kjellander@webrtc.org · 10 years ago
  18. 068cd6f Fix failing test introduced with r6111. by stefan@webrtc.org · 10 years ago
  19. 4d61a36 Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. by stefan@webrtc.org · 10 years ago
  20. 5435208 Upping start bitrate to min, if set to a lower value i SetSendCodec. by mflodman@webrtc.org · 10 years ago
  21. c5fccd6 Disable flaky RunsRtpRtcpTestWIthoutErrors. by pbos@webrtc.org · 10 years ago
  22. ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
  23. 9d10769 Propagate capture ntp timestamp from rtp to renderer. by wu@webrtc.org · 10 years ago
  24. 98f8320 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. by fischman@webrtc.org · 10 years ago
  25. 022615b Log Fixit for parts of video_engine folder. by mflodman@webrtc.org · 10 years ago
  26. 2f0c5f7 Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. by solenberg@webrtc.org · 10 years ago
  27. 2d3624c Implement ViE forwarding to RBE of packets for BWE coming in through the ViENetwork::ReceivedBWEPacket API. by solenberg@webrtc.org · 10 years ago
  28. a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
  29. 800136d Remove ViE external encryption API. by solenberg@webrtc.org · 10 years ago
  30. fba4f1c Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
  31. c9faf10 Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  32. aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  33. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  34. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  35. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  36. a48c91d Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
  37. ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  38. cebd1d7 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
  39. ee234be Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  40. a4fae33 Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
  41. db43763 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. by dwkang@webrtc.org · 11 years ago
  42. 82b883c Renaming ViEEncoderObserver::VideoSuspended by henrik.lundin@webrtc.org · 11 years ago
  43. 3b7da1e Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. by asapersson@webrtc.org · 11 years ago
  44. 63e3810 Typo in vie_autotest_win.cc by braveyao@webrtc.org · 11 years ago
  45. 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  46. 4590177 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  47. 2f9e587 Disable all vie_auto_tests on Linux for now (take 2) by kjellander@webrtc.org · 11 years ago
  48. 8167387 Disable all automated vie_auto_tests on Linux for now by kjellander@webrtc.org · 11 years ago
  49. 24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
  50. 4ce7590 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
  51. ecfef19 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  52. 6036f56 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
  53. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  54. 7c46e95 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
  55. 63301bd Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  56. c5b5ad1 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
  57. 51e0101 Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  58. 44bb62a Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
  59. 6c9c551 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
  60. a24c356 Run FullStack tests without render windows. by pbos@webrtc.org · 11 years ago
  61. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  62. e2c52d7 Move ChromaGenerator to common_video/. by pbos@webrtc.org · 11 years ago
  63. 9caedd0 Android: Fixes WebRTCDemo build (missing Java code). by henrike@webrtc.org · 11 years ago
  64. 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
  65. bf1da46 Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
  66. 4b14e5a Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
  67. 81cd5ca VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
  68. 22a2893 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
  69. b5d2d16 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
  70. 39079d1 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
  71. c5080a9 Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
  72. 362e3e5 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
  73. a89f7e8 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
  74. 890706b Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
  75. b0382ea Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
  76. 199555c Revert test change in r4808. by stefan@webrtc.org · 11 years ago
  77. d704640 Reduce flakiness in network down test. by stefan@webrtc.org · 11 years ago
  78. 0011252 Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
  79. 28a1166 Rename EngineTest to CallTest. by pbos@webrtc.org · 11 years ago
  80. 28631e7 Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 11 years ago
  81. a89566f Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 11 years ago
  82. 93b9912 Fixes a flake in network down tests. by stefan@webrtc.org · 11 years ago
  83. 041d54b Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
  84. bfad17e Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  85. 990c5e3 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
  86. f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
  87. eb2d9dd Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 11 years ago
  88. fa996f2 Split up EngineTests and RampupTests. by pbos@webrtc.org · 11 years ago
  89. bf6d572 Rename VideoCall to Call. by pbos@webrtc.org · 11 years ago
  90. 618a0ec ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
  91. ca20f3d Clamp camera id to legal values. by fischman@webrtc.org · 11 years ago
  92. db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
  93. 0020858 Remove send and receive streams when destroyed. by pbos@webrtc.org · 11 years ago
  94. 4998966 Allow unknown flags in test_main.cc. by pbos@webrtc.org · 11 years ago
  95. c77dcb0 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. by mflodman@webrtc.org · 11 years ago
  96. 1cd055c Disable EngineTest.ReceivesPliAndRecoversWithNack. by mflodman@webrtc.org · 11 years ago
  97. 9e70940 Add FakeEncoder to VideoSendStream tests. by pbos@webrtc.org · 11 years ago
  98. 8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
  99. 9b7bdee Revert r4562 by elham@webrtc.org · 11 years ago
  100. e2e033a Relanding 4597 - Don't force key frame when decoding with errors. by mikhal@webrtc.org · 11 years ago