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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 0d03514a1a95cd83126ba35102c4277555603577 / video_engine
  1. d1e7fac Add stats of incoming frame delays for debugging bandwidth estimation. by jiayl@webrtc.org · 10 years ago
  2. d2f95a8 Connect webrtc::Config to WrappingBitrateEstimator by henrik.lundin@webrtc.org · 11 years ago
  3. 7dc7514 Revert 5444 "Revert 5421 "Fix deadlock on register/unregister ob..." by mallinath@webrtc.org · 11 years ago
  4. 9afb6cf Revert 5421 "Fix deadlock on register/unregister observer while ..." by mallinath@webrtc.org · 11 years ago
  5. 630939f Fix deadlock on register/unregister observer while there is a an going callback. by andresp@webrtc.org · 11 years ago
  6. 7d99cd4 Add callbacks for receive channel RTP statistics by sprang@webrtc.org · 11 years ago
  7. b4263e0 Add configuration and test for extended RTCP reference time reports to new video api. by asapersson@webrtc.org · 11 years ago
  8. fba4f1c Roll Chromium 238260 -> 243863 by wjia@webrtc.org · 11 years ago
  9. eed1f11 Updated Webrtc version to 3.49 by elham@webrtc.org · 11 years ago
  10. 083049f Removes usage of ListWrapper from several files. by henrike@webrtc.org · 11 years ago
  11. 49812e6 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
  12. c9faf10 Add thread_annotations for clang targets. by andresp@webrtc.org · 11 years ago
  13. a9a7327 If the configured start bitrate is higher than the configures max by mflodman@webrtc.org · 11 years ago
  14. 9662535 Race condition in ViECapturer::RegisterObserver by sprang@webrtc.org · 11 years ago
  15. 91cebfc Update WebRTC to version 3.48 by tnakamura@webrtc.org · 11 years ago
  16. 4f1f5fa Add callbacks for receive channel RTCP statistics. by sprang@webrtc.org · 11 years ago
  17. aacdb9f Integrate fake_network_pipe into direct_transport. by stefan@webrtc.org · 11 years ago
  18. 5596ac6 Remove media_file from VideoEngine dependencies. by pbos@webrtc.org · 11 years ago
  19. 46f7288 Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
  20. c5a5713 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
  21. 3bbc91e Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
  22. b70db6d Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
  23. dadfc9e Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
  24. c49a3fa Making RemoteRateControl::min_configured_bit_rate_ configurable by henrik.lundin@webrtc.org · 11 years ago
  25. efeb8ce Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  26. 12553ad Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  27. 984bee2 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  28. a48c91d Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
  29. 4fcb2f5 Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
  30. c33d37c Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  31. ffea4ce Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  32. ca63ad9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  33. 1430bc3 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  34. cebd1d7 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
  35. b113981 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  36. ee234be Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  37. a4fae33 Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
  38. db04941 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  39. d964bf5 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
  40. 9b30fd3 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  41. c7d7363 Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  42. 1fc02eb Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
  43. 5fdd10a Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  44. f1630b1 Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second. by asapersson@webrtc.org · 11 years ago
  45. 47f0c41 Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  46. 5a669d5 Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
  47. 04d6593 Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
  48. 163393e Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
  49. 2e98d45 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  50. dd4f866 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
  51. 4ee440a Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
  52. b0d97ed Updated WebRTC version to 3.47 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  53. b1d7931 Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
  54. 50293f5 Replace VideoFrameI420 with I420VideoFrame. by pbos@webrtc.org · 11 years ago
  55. db43763 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. by dwkang@webrtc.org · 11 years ago
  56. 82b883c Renaming ViEEncoderObserver::VideoSuspended by henrik.lundin@webrtc.org · 11 years ago
  57. 7673871 Protect reads of ViEEncoder::video_suspended_. by pbos@webrtc.org · 11 years ago
  58. b9f1eb8 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  59. 3dc7ff3 Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  60. 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
  61. 63e3810 Typo in vie_autotest_win.cc by braveyao@webrtc.org · 11 years ago
  62. 4673674 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  63. 4747585 Added ViE API for getting overuse measure. by asapersson@webrtc.org · 11 years ago
  64. 3051951 Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
  65. c4af4cf Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). by asapersson@webrtc.org · 11 years ago
  66. 4590177 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  67. 2f9e587 Disable all vie_auto_tests on Linux for now (take 2) by kjellander@webrtc.org · 11 years ago
  68. 8167387 Disable all automated vie_auto_tests on Linux for now by kjellander@webrtc.org · 11 years ago
  69. b748c9d Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
  70. 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
  71. 78726d1 Updated WebRTC version to 3.46 by elham@webrtc.org · 11 years ago
  72. f4def77 Sending status fix for module. by asapersson@webrtc.org · 11 years ago
  73. 1bd9a7b Removed unused code. by asapersson@webrtc.org · 11 years ago
  74. af92d3e Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  75. a191cb0 Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  76. 6baaf30 Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" by sheu@chromium.org · 11 years ago
  77. 7773eec Remove extra copy in VideoCaptureImpl::IncomingFrameI420 by sheu@chromium.org · 11 years ago
  78. 6646abd Video bandwidth not reported correctly by sprang@webrtc.org · 11 years ago
  79. f00942a Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. by fischman@webrtc.org · 11 years ago
  80. 24e2089 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago
  81. 4ce7590 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
  82. ecfef19 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. by fischman@webrtc.org · 11 years ago
  83. 6036f56 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
  84. 221798a Upgrade scoped_ptr to Chromium's latest version. by andrew@webrtc.org · 11 years ago
  85. 6796d68 Updated WebRTC version to 3.45 by elham@webrtc.org · 11 years ago
  86. 4633e15 Changing the bitrate clamping in BitrateControllerImpl by henrik.lundin@webrtc.org · 11 years ago
  87. 7c46e95 AutoMute: Adding channel_id parameter to callback. by henrik.lundin@webrtc.org · 11 years ago
  88. 63301bd Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
  89. c5b5ad1 Make sure the first frame isn't dropped. by pbos@webrtc.org · 11 years ago
  90. 5e74d96 Have padding decay to zero if no frames are being captured. by stefan@webrtc.org · 11 years ago
  91. 51e0101 Compound/reduced-size RTCP in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  92. 44bb62a Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
  93. 93cd397 Don't pad if only one stream is sent, except if auto muted. by stefan@webrtc.org · 11 years ago
  94. 6c9c551 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
  95. a24c356 Run FullStack tests without render windows. by pbos@webrtc.org · 11 years ago
  96. 9653397 Roll chromium_revision 226126:228675 and fix clang warnings by kjellander@webrtc.org · 11 years ago
  97. e2c52d7 Move ChromaGenerator to common_video/. by pbos@webrtc.org · 11 years ago
  98. 9caedd0 Android: Fixes WebRTCDemo build (missing Java code). by henrike@webrtc.org · 11 years ago
  99. 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
  100. eeaea08 Updated WebRTC version to 3.44 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago