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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / c1797061756f40b9b1f3f3e82fc040ce18ecd43c / video_engine
  1. c179706 Remove newapi:: namespace for typenames without overlap. by pbos@webrtc.org · 11 years ago
  2. f83a872 Revert 4597 "Don't force key frame when decoding with errors" by henrike@webrtc.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. a20e2d4 Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" by stefan@webrtc.org · 11 years ago
  9. 3ded8c9 Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots. by henrike@webrtc.org · 11 years ago
  10. c0976d2 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." by henrike@webrtc.org · 11 years ago
  11. efe1f0f Reverts a second set of reverts caused by a bug in a dependency. by stefan@webrtc.org · 11 years ago
  12. f96e534 Call SetExecutablePath from test_main.cc by pbos@webrtc.org · 11 years ago
  13. 7deb335 Make FrameGeneratorCapturer own frame_generator. by pbos@webrtc.org · 11 years ago
  14. eb7b0c4 Merging video_full_stack_tests and video_engine_tests. by phoglund@webrtc.org · 11 years ago
  15. 67acd69 VideoSendStream SSRC test. by pbos@webrtc.org · 11 years ago
  16. 96ff6ab Added missing static_cast conversion. by pbos@webrtc.org · 11 years ago
  17. 8ce445e Implementation and testing of PLI in new API. by pbos@webrtc.org · 11 years ago
  18. 3207eaa Made all integration tests use consistent naming. by phoglund@webrtc.org · 11 years ago
  19. ece3d35 Added choice of decode error mode to loopback test. by agalusza@google.com · 11 years ago
  20. 7fc75bb Update talk to 50918584. by wu@webrtc.org · 11 years ago
  21. 1e817c3 Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 11 years ago
  22. e155918 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
  23. 298bbdb Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
  24. e416ab2 Remove ViEBase::Init() call from VideoCall. by pbos@webrtc.org · 11 years ago
  25. c2014fd Remove VideoEngine class from new VideoEngine API. by pbos@webrtc.org · 11 years ago
  26. eca72bf Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors". by marpan@webrtc.org · 11 years ago
  27. 48bcf6f Disable racy part of RunsRtpRtcpTestWithoutErrors. by pbos@webrtc.org · 11 years ago
  28. 52c5c70 Replace MapWrapper with std::map<>. by pbos@webrtc.org · 11 years ago
  29. 8c8c87f Updated WebRTC version to 3.39 by elham@webrtc.org · 11 years ago
  30. 823a888 Signal when shutting down DirectTransport. by pbos@webrtc.org · 11 years ago
  31. d893b3f Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  32. fe881f6 Run loopback tests with network thread. by pbos@webrtc.org · 11 years ago
  33. f43029b Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 11 years ago
  34. b0af417 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  35. e915174 Allowing decoding with errors, when disabling nack. by mikhal@webrtc.org · 11 years ago
  36. ea7b33e * Update libjingle to 50389769. by wu@webrtc.org · 11 years ago
  37. 3ddbca9 Updated WebRTC version number to 3.38 by elham@webrtc.org · 11 years ago
  38. 3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  39. 043f6a8 Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. by pbos@webrtc.org · 11 years ago
  40. 78ab511 Use RtpHeaderParser in VideoCall implementation. by pbos@webrtc.org · 11 years ago
  41. ce85109 Glue code and tests for NACK in new VideoEngine API. by pbos@webrtc.org · 11 years ago
  42. 3a74d40 Fix send times in video_full_stack. by pbos@webrtc.org · 11 years ago
  43. 8704595 Add back is.FrameProvider() call lost in r4194. by pbos@webrtc.org · 11 years ago
  44. acb00f5 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  45. aa79e6e Unbreak clang/android build of webrtc. by fischman@webrtc.org · 11 years ago
  46. cb9a72b Adding possibility to use encoding time when trigger underuse for frame based overuse detection. by mflodman@webrtc.org · 11 years ago
  47. 45e69ce Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
  48. d69e2f4 Access receiving_ under receive_cs critical section by braveyao@webrtc.org · 11 years ago
  49. bf76ae2 Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago
  50. 237fd31 Correctly rebuild WebRTCDemo after jni/ source file changes by yujie.mao@webrtc.org · 11 years ago
  51. 0ba496b Revert r4301 by tnakamura@webrtc.org · 11 years ago
  52. b89eed3 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on by elham@webrtc.org · 11 years ago
  53. 5526bfb Updated WebRTC version to 3.36 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
  54. e835019 Include files from webrtc/.. paths in video_engine/. by pbos@webrtc.org · 11 years ago
  55. 778a172 Direct3D renderer for new VideoEngine API tests. by pbos@webrtc.org · 11 years ago
  56. 46088d2 Support sending multiple report blocks and keeping track of statistics on several SSRCs. by stefan@webrtc.org · 11 years ago
  57. 73a2ba6 Update version number to 3.35 by tnakamura@webrtc.org · 11 years ago
  58. 44cff54 Update version number to 3.34 by tnakamura@webrtc.org · 11 years ago
  59. e9bd299 Fixed implicit-int-conversion bugs. by pbos@webrtc.org · 11 years ago
  60. 57dbdbd Fix a circular dependency by removing an unnecessary dependency, add a missing include_tests check and missing lib references for android. by stefan@webrtc.org · 11 years ago
  61. 2f02da8 Initial port of FullStackTest to new VideoEngine API. by pbos@webrtc.org · 11 years ago
  62. 9c0f14d Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
  63. 7e27a0f Fixes build breakage when building WebRTC in Chromium and having include_tests=1. by henrike@webrtc.org · 11 years ago
  64. b16cb05 Use scoped_ptr<> for loopback.cc by pbos@webrtc.org · 11 years ago
  65. a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
  66. 39445b0 Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
  67. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  68. 438be80 Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
  69. 50c1aef Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  70. 5e742a8 WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
  71. c7eab28 Re-add WebRTCDemo dependencies as dependencies (not just inputs) because they also need to be built for this target! by fischman@webrtc.org · 11 years ago
  72. 555f1cd Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
  73. f8f91d6 Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
  74. 0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
  75. 20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  76. 0f6f7cb Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
  77. 7533659 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
  78. 69f7605 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
  79. b06dd93 Fix AV sync issue by hclam@chromium.org · 11 years ago
  80. 9540e2a Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
  81. 6828566 WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
  82. cefb004 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
  83. 8ed5369 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  84. 83163e0 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  85. f09f7b2 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  86. 46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  87. 0604490 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  88. f1bcae0 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
  89. b4c89a4 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
  90. 63988b2 RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
  91. a18c6e5 Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
  92. 6fb2ca3 Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
  93. 6f1c3ef Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
  94. d8ecee5 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
  95. e54928f Remove XvRenderer. by pbos@webrtc.org · 11 years ago
  96. 695ff2a Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
  97. 0016110 Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
  98. e3e4615 Use int for FPS instead of size_t. by pbos@webrtc.org · 11 years ago
  99. 54b6ebc Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
  100. 23e3f44 Remove assert for aborting FrameGeneratorCapturer. by pbos@webrtc.org · 11 years ago