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gerrit-public.fairphone.software / platform / external / webrtc / 2018269dc3a1c1bb01c946583ca0750ae0db68e3
  1. 2018269 Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  2. a129b6c Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  3. 451745e Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
  4. 88ac63a Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
  5. df7b1d6 AppRTCDemo(android): make ant be quiet on success and not overly noisy on failure. by fischman@webrtc.org · 11 years ago
  6. 9ee75e9 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency). by henrike@webrtc.org · 11 years ago
  7. f41f06b PeerConnection(java): rationalize pointer-to-jlong conversion. by fischman@webrtc.org · 11 years ago
  8. 9caf276 Update talk to 58037405. by wu@webrtc.org · 11 years ago
  9. 391b4db Fix common_video_unittests in apk_tests.gyp. by pbos@webrtc.org · 11 years ago
  10. 724947b Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  11. 4c3faa9 Disable a libjingle unittest which is failing after a chromium roll out. by turaj@webrtc.org · 11 years ago
  12. df02283 Adds audio volume demo to the index page. by hta@webrtc.org · 11 years ago
  13. 59d5705 Fix memory tools error introduced in roll @ r5260 by kjellander@webrtc.org · 11 years ago
  14. 096e8d9 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  15. f9bdbe3 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  16. 2656cf9 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  17. 26c40ba Removed audio element from volume measuring demo. by hta@webrtc.org · 11 years ago
  18. 1133ffd Merged OWNERS of JS demo directories by hta@webrtc.org · 11 years ago
  19. c4038d7 Rewriting the SoundMeter class to be RMS and be encapsulated differently by hta@webrtc.org · 11 years ago
  20. 77507ef Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
  21. 7ae8495 Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
  22. 762fcdc Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
  23. 8b88192 Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
  24. 917306d Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
  25. eb7def2 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  26. c329529 Apply transaction to setting connected to Room entities, to resolve a possible race condition at two clients connecting simultaneously. by braveyao@webrtc.org · 11 years ago
  27. 70ddf93 libyuv r905 with yuv off by 1 fix for valgrind overread by fbarchard@google.com · 11 years ago
  28. de7c9e8 Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago
  29. 5e13ac9 Add shape in DesktopFrame. by sergeyu@chromium.org · 11 years ago
  30. 4acf450 libyuv roll to r888 with valgrind overread fixes. by fbarchard@google.com · 11 years ago
  31. 8d0ca7f Add new method to MockAudioProcessing. by andrew@webrtc.org · 11 years ago
  32. 797522f Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
  33. 863b536 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  34. 0f3d0bb Stop video capturers in multi-stream test. by pbos@webrtc.org · 11 years ago
  35. 758db4b Demo showing how to measure volume using WebAudio by hta@webrtc.org · 11 years ago
  36. 88615f0 Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks by sprang@webrtc.org · 11 years ago
  37. 7f73280 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
  38. 32f485b Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  39. 57a5f64 revert r5230 by sergeyu@chromium.org · 11 years ago
  40. a1b21cd Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  41. 7104fc1 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered. by sprang@webrtc.org · 11 years ago
  42. 96a9b2d Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. by asapersson@webrtc.org · 11 years ago
  43. ebad765 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  44. 5cea89f Remove CallTest dependency on voice_engine/test/. by pbos@webrtc.org · 11 years ago
  45. 0a3c147 Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  46. 07fcc4f Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
  47. c49d5b7 Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
  48. 245037d Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  49. b88fc18 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
  50. a6ad6e5 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  51. c4726d0 Make RTPSender::SendPadData public. by stefan@webrtc.org · 11 years ago
  52. 5bc25c4 Update libjingle to 57692857 by sergeyu@chromium.org · 11 years ago
  53. 3d9981d Remove unused ThreadData struct. by andrew@webrtc.org · 11 years ago
  54. 3054ba6 Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  55. 5b51ebc Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
  56. 71f055f Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  57. 9e5b034 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
  58. 79b6320 Fixes a crash in fullstack tests introduced with r5209. by stefan@webrtc.org · 11 years ago
  59. b477fa6 Small fixes to plot_neteq_delay.m by henrik.lundin@webrtc.org · 11 years ago
  60. 7e9315b Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  61. 9523b55 Fix a typo in neteq.gypi by henrik.lundin@webrtc.org · 11 years ago
  62. d7696c4 Compile-out functions only used by the bit-exact test. by andrew@webrtc.org · 11 years ago
  63. d3865e9 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). by fischman@webrtc.org · 11 years ago
  64. 812dd11 Add baseline generation/verification to BWE test framework. by solenberg@webrtc.org · 11 years ago
  65. 499631c Utility class for reading/writing network-byte-ordered integers. by sprang@webrtc.org · 11 years ago
  66. 37968a9 Change BitrateStats to more generalized RateStatistics by sprang@webrtc.org · 11 years ago
  67. b613b5a Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  68. 5ecdef1 Do not use recursive calling in NetEq test tools by henrik.lundin@webrtc.org · 11 years ago
  69. e003455 RTCPeerConnection(objc): avoid leaking ICE candidate on addition. by fischman@webrtc.org · 11 years ago
  70. 8418e96 Fixing NetEq tests for new Opus version by tina.legrand@webrtc.org · 11 years ago
  71. 54e8bfa Apprtc demo: add DSCP support. by braveyao@webrtc.org · 11 years ago
  72. 03c7a35 Fixing long lines in apprtc.py. by phoglund@webrtc.org · 11 years ago
  73. e1fc3f2 Disable check for all sent SSRCs being valid. by pbos@webrtc.org · 11 years ago
  74. bd41a84 This CL adds an API to enable robust validation of delay estimates. by bjornv@webrtc.org · 11 years ago
  75. b627f67 Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large. by stefan@webrtc.org · 11 years ago
  76. 1f7c8d8 Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
  77. 13d38a1 Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
  78. d1a1c35 Recommit CL5184 by bjornv@webrtc.org · 11 years ago
  79. c8f76dd Refactor Remote Estimators Test into a more reusable form. by solenberg@webrtc.org · 11 years ago
  80. 82eb3a6 Revert 5184 "Small refactoring change in delay_estimator." by bjornv@webrtc.org · 11 years ago
  81. eea079a Small refactoring change in delay_estimator. by bjornv@webrtc.org · 11 years ago
  82. 19a40ff Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
  83. b3ea3af Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
  84. 4070935 Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  85. c7ff8f9 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
  86. bd51d93 LSan suppressions for libjingle_peerconnection_unittest by kjellander@webrtc.org · 11 years ago
  87. 7f95998 Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
  88. d89b52a Faster implementation of BitRateStats. by mikhal@webrtc.org · 11 years ago
  89. 326bcff Updated WebRTC version to 3.47 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  90. 4e3161d Style-option file for clang-format. by pbos@webrtc.org · 11 years ago
  91. 3260f10 Made video quality toolchain more configurable. by phoglund@webrtc.org · 11 years ago
  92. 47fadba Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
  93. 4ab4fc0 Add test for automatically disabling padding when no video is being captured. by stefan@webrtc.org · 11 years ago
  94. b5bc098 Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error. by fbarchard@google.com · 11 years ago
  95. aa74b5d Add success/error callback to set sdp calls. by wu@webrtc.org · 11 years ago
  96. 5272eb8 Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. by turaj@webrtc.org · 11 years ago
  97. e839da0 Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. by sergeyu@chromium.org · 11 years ago
  98. 78b41a0 Fix issues with sequence number wrap-around in jitter statistics. by turaj@webrtc.org · 11 years ago
  99. 832bd74 libyuv r874 for build improvements on ios/android, and improved YUV scale performance. by fbarchard@google.com · 11 years ago
  100. b43202d Disable PeerConnectionEndToEndTest for tsanv2 build. by wu@webrtc.org · 11 years ago