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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 00d566ec297d52c8e732d7299d69e4e580707d61 / video_engine
  1. 39445b0 Adding Stefan as VideoEngine owner, removing Per. by mflodman@webrtc.org · 11 years ago
  2. 3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
  3. 438be80 Remove log of undefined input values in GetCodec. by pbos@webrtc.org · 11 years ago
  4. 50c1aef Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  5. 5e742a8 WebRTCViEDemo: Use global reference when passing variables across different threads by yujie.mao@webrtc.org · 11 years ago
  6. 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
  7. 555f1cd Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes. by fischman@webrtc.org · 11 years ago
  8. f8f91d6 Adding a first simple version of overuse detection, but not hooked up. by mflodman@webrtc.org · 11 years ago
  9. 0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
  10. 20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
  11. 0f6f7cb Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
  12. 7533659 Fixes some pacer/padding issues found while testing. by stefan@webrtc.org · 11 years ago
  13. 69f7605 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
  14. b06dd93 Fix AV sync issue by hclam@chromium.org · 11 years ago
  15. 9540e2a Log current and target AV delay in ViESyncModule by hclam@chromium.org · 11 years ago
  16. 6828566 WebRTCDemo: ensures that using front and back camera work as expected. by henrike@webrtc.org · 11 years ago
  17. cefb004 Revert 4211 "Build all java files into jar for each module on An..." by fischman@webrtc.org · 11 years ago
  18. 8ed5369 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  19. 83163e0 Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  20. f09f7b2 Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test. by kjellander@webrtc.org · 11 years ago
  21. 46cec2a Reorganize test targets in WebRTC by kjellander@webrtc.org · 11 years ago
  22. 0604490 Build all java files into jar for each module on Android by fischman@webrtc.org · 11 years ago
  23. f1bcae0 Updated WebRTC version to 3.33 by elham@webrtc.org · 11 years ago
  24. b4c89a4 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
  25. 63988b2 RW lock access to ssrc maps in VideoCall. by pbos@webrtc.org · 11 years ago
  26. a18c6e5 Removing functionality for inserting pre-encoded frames instead of raw by mflodman@webrtc.org · 11 years ago
  27. 6fb2ca3 Fix init list for VideoSendStream::Config::Rtp. by pbos@webrtc.org · 11 years ago
  28. 6f1c3ef Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
  29. d8ecee5 Update the remote bitrate estimator before passing the packet to the RTP module. by stefan@webrtc.org · 11 years ago
  30. e54928f Remove XvRenderer. by pbos@webrtc.org · 11 years ago
  31. 695ff2a Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
  32. 0016110 Setting SSRC in vie_loopback_test by mikhal@webrtc.org · 11 years ago
  33. e3e4615 Use int for FPS instead of size_t. by pbos@webrtc.org · 11 years ago
  34. 54b6ebc Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
  35. 23e3f44 Remove assert for aborting FrameGeneratorCapturer. by pbos@webrtc.org · 11 years ago
  36. c1506a2 Fake VideoCapturer based on FrameGenerator by pbos@webrtc.org · 11 years ago
  37. 4e5f983 Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
  38. 6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
  39. 4988d94 Break video_engine/new_include/common.h into smaller parts. by pbos@webrtc.org · 11 years ago
  40. 5221d1c Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp. by andrew@webrtc.org · 11 years ago
  41. 3990df2 Updated WebRTC version to 3.32 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  42. e3b52e6 Don't return an estimated receive BW for channels not receiving video. by mflodman@webrtc.org · 11 years ago
  43. bb6bef5 Include gflags with "gflags/gflags.h" instead of <> by pbos@webrtc.org · 11 years ago
  44. 39784c4 Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness. by stefan@webrtc.org · 11 years ago
  45. b2d1a40 Default constructors for new VideoEngine structs. by pbos@webrtc.org · 11 years ago
  46. 5437a2c Remove libvpx_intrinsics_sse4_1.a in Android.mk since this target is no longer generated in libvpx by fischman@webrtc.org · 11 years ago
  47. f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
  48. a93cbbf Adding Mac test renderer, some test refactoring and made cpplint pass. by mflodman@webrtc.org · 11 years ago
  49. 460e172 Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc. by stefan@webrtc.org · 11 years ago
  50. ad6cade Make sure GlxRenderer frees its resources. by pbos@webrtc.org · 11 years ago
  51. eef4fd5 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
  52. 8f1d1a9 Fix regression where retransmission bitrate is no longer estimated. by stefan@webrtc.org · 11 years ago
  53. 42e1fe1 CreateEmptyFrame casts from size_t to int. by pbos@webrtc.org · 11 years ago
  54. 08f3ca9 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
  55. 074eb20 Control new VideoEngine tests with gflags. by pbos@webrtc.org · 11 years ago
  56. b59962f Adds print out of incoming resolution. by henrike@webrtc.org · 11 years ago
  57. d557734 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
  58. b7716d8 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. by solenberg@webrtc.org · 11 years ago
  59. 141a00c Remove <iostream> usage from loopback.cc by pbos@webrtc.org · 11 years ago
  60. 6169712 Suffix VcmCapturer's privates with underscore_ by pbos@webrtc.org · 11 years ago
  61. cca5086 Log error in ViESender::SendRTCPPacket by hclam@chromium.org · 11 years ago
  62. 453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
  63. b9e5732 Avoid NPE crash on Android platforms that don't support getting preview framerate. by fischman@webrtc.org · 11 years ago
  64. 9ea8c99 Include gflags properly and X11 include order in VideoEngine. by pbos@webrtc.org · 11 years ago
  65. 281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
  66. e8dc588 Enable WebRTC demo application on x86 Android by fischman@webrtc.org · 11 years ago
  67. 5a22c40 Cleanup traces in WebRTC by hclam@chromium.org · 11 years ago
  68. 2a9108f New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago
  69. 7645e4d Remove SetOverUseDetectorOptions and cleaned ViESharedData. by mflodman@webrtc.org · 11 years ago
  70. 0425392 Adding a factory to remote bitrate estimator and allow it to be set via config. by andresp@webrtc.org · 11 years ago
  71. fb20e53 Fix typo in log statement. witdh should be width. by fbarchard@google.com · 11 years ago
  72. d474c13 Add more tracing for key frames. by justinlin@chromium.org · 11 years ago
  73. dcfeff7 Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343. by vikasmarwaha@webrtc.org · 11 years ago
  74. 08fe40f Updated WebRTC version to 3.31 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  75. 9062a9a Disabled flaky codec test (RunsCodecTestWithoutErrors) by phoglund@webrtc.org · 11 years ago
  76. ac6d919 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
  77. 7d6e2a0 Remove TEXT(x) for BUILDINFO macros. by pbos@webrtc.org · 11 years ago
  78. 15c0af4 This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build. by fischman@webrtc.org · 11 years ago
  79. f22bfed Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation. by fischman@webrtc.org · 11 years ago
  80. af89cc3 WebRTCDemo Android doesn't hangle activity recreation correctly. by braveyao@webrtc.org · 11 years ago
  81. 4afe86b Add fischman into OWNERS of WebRTCDemo Android. by braveyao@webrtc.org · 11 years ago
  82. 7ab7268 Fix compile errors in ViE with latest clang. by andrew@webrtc.org · 11 years ago
  83. 90f05ed Clean creation of VideoEngine: by andresp@webrtc.org · 11 years ago
  84. 06ad384 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 11 years ago
  85. 98b2011 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 11 years ago
  86. b6c7447 Updated WebRTC version number to 3.30 by elham@webrtc.org · 11 years ago
  87. 3a3a787 Get rid of some unnecessary copying when sending REMBs. by solenberg@webrtc.org · 11 years ago
  88. 677ca88 Removing bad code resulting in flaky test. by pwestin@webrtc.org · 11 years ago
  89. 36bdba4 Adding trace and changing pacing constants by pwestin@webrtc.org · 11 years ago
  90. 422a124 Bugfix custom call stop. by pwestin@webrtc.org · 11 years ago
  91. be6ad88 WebRTCDemo Android app to route audio to headphone when it's plugged in. by braveyao@webrtc.org · 11 years ago
  92. 166153e Consolidate common_audio into a single target. by andrew@webrtc.org · 11 years ago
  93. 4d4ba47 Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync. by pwestin@webrtc.org · 11 years ago
  94. 6ed1c8c Adding buffered mode to loopback test by mikhal@webrtc.org · 11 years ago
  95. 1a58dd7 Removing vie file related code from vie_custom_call by mikhal@webrtc.org · 11 years ago
  96. b13f394 Fixed remaining nits from Stefan by pwestin@webrtc.org · 11 years ago
  97. 3816c52 Fix the encoder pause logic. BUG=1691 by pwestin@webrtc.org · 11 years ago
  98. 4eac481 Disabling avi file interface by mikhal@webrtc.org · 11 years ago
  99. d430f32 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
  100. 2788107 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago