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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 3a4fc4bb80bcc948642315daf07bc21fd93a52ab
  1. 3a4fc4b Update talk to 58127566 together with by wu@webrtc.org · 11 years ago
  2. b9cf1de ACM 2 compatibility with ACM 1. by turaj@webrtc.org · 11 years ago
  3. 9b3d2bf Revert 5274 "Update talk to 58113193 together with https://webrt..." by wu@webrtc.org · 11 years ago
  4. cde78d6 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. by wu@webrtc.org · 11 years ago
  5. a4670a1 Complete rewrite of demo application. by henrike@webrtc.org · 11 years ago
  6. 0ceb51f Remove overloaded CpuOveruseMeasure function. by asapersson@webrtc.org · 11 years ago
  7. 2f70422 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
  8. f49d16c Fix common_video_unittests in apk_tests.gyp. by pbos@webrtc.org · 11 years ago
  9. 7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
  10. 66e84b0 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
  11. 894dab9 Roll chromium_revision 232627:238260 by kjellander@webrtc.org · 11 years ago
  12. f1d22d4 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
  13. 72b0d40 Removed unnecessary Pulse init from VoE startup. by fischman@webrtc.org · 11 years ago
  14. e8ca064 Correctly define OVERRIDE when building with g++ 4.7 and C++11 support by andrew@webrtc.org · 11 years ago
  15. 090f37f Improve VideoSendStreamTest::MaxPacketSize by sprang@webrtc.org · 11 years ago
  16. ba8b32c Change uses of the obsolete armv7 setting to arm_version==7. by kjellander@webrtc.org · 11 years ago
  17. 934be30 Fix compilation errors on Fedora 20. by fischman@webrtc.org · 11 years ago
  18. 4adc7ad Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0. by andrew@webrtc.org · 11 years ago
  19. e681a01 Add shape in DesktopFrame. by sergeyu@chromium.org · 11 years ago
  20. e8dd108 Add new method to MockAudioProcessing. by andrew@webrtc.org · 11 years ago
  21. e4d591a Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." by andrew@webrtc.org · 11 years ago
  22. c8bd975 Allow opening an AEC dump from an existing file handle. by henrikg@webrtc.org · 11 years ago
  23. 9e40eba Stop video capturers in multi-stream test. by pbos@webrtc.org · 11 years ago
  24. 5b23ce6 Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks by sprang@webrtc.org · 11 years ago
  25. ed8c496 Fraction lost statistics not being reported by sprang@webrtc.org · 11 years ago
  26. 6dccf13 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
  27. 2de68d6 Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. by asapersson@webrtc.org · 11 years ago
  28. cf5c552 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
  29. f0d9b20 Remove CallTest dependency on voice_engine/test/. by pbos@webrtc.org · 11 years ago
  30. 8db148e Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
  31. adc238a Fixing the android build by henrik.lundin@webrtc.org · 11 years ago
  32. 3d70641 Move implementation files out of the webrtc/ root. by pbos@webrtc.org · 11 years ago
  33. b669e60 Remove default implementations for SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  34. 3bcea52 Fix bug where fraction_lost is always set to 0 when getting received RTCP statistics. by stefan@webrtc.org · 11 years ago
  35. 8911937 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
  36. d90bebb Make RTPSender::SendPadData public. by stefan@webrtc.org · 11 years ago
  37. 991d58c Remove unused ThreadData struct. by andrew@webrtc.org · 11 years ago
  38. 5459e0b Remove the long disabled WEBRTC_SVNREVISION define. by andrew@webrtc.org · 11 years ago
  39. 382cfdd Removing DropDeltaAfterKey functionality which is unused. by andresp@webrtc.org · 11 years ago
  40. 9435a17 Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
  41. f2c136b 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
  42. 0cb8020 Fixes a crash in fullstack tests introduced with r5209. by stefan@webrtc.org · 11 years ago
  43. 29a9669 Small fixes to plot_neteq_delay.m by henrik.lundin@webrtc.org · 11 years ago
  44. da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
  45. c101a27 Fix a typo in neteq.gypi by henrik.lundin@webrtc.org · 11 years ago
  46. c749348 Compile-out functions only used by the bit-exact test. by andrew@webrtc.org · 11 years ago
  47. b1f4a72 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). by fischman@webrtc.org · 11 years ago
  48. adaa8b5 Add baseline generation/verification to BWE test framework. by solenberg@webrtc.org · 11 years ago
  49. 4ed6832 Utility class for reading/writing network-byte-ordered integers. by sprang@webrtc.org · 11 years ago
  50. 397aae0 Change BitrateStats to more generalized RateStatistics by sprang@webrtc.org · 11 years ago
  51. 309b2c8 Set local SSRC for VideoReceiveStream. by pbos@webrtc.org · 11 years ago
  52. f91a14f Do not use recursive calling in NetEq test tools by henrik.lundin@webrtc.org · 11 years ago
  53. 02817f8 Fixing NetEq tests for new Opus version by tina.legrand@webrtc.org · 11 years ago
  54. 169a27a Disable check for all sent SSRCs being valid. by pbos@webrtc.org · 11 years ago
  55. 758ef4c This CL adds an API to enable robust validation of delay estimates. by bjornv@webrtc.org · 11 years ago
  56. c8918cb 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
  57. 0e2571d Lock frame in ViECapturer::IncomingFrameI420. by pbos@webrtc.org · 11 years ago
  58. 9105cbd Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
  59. b8b2a23 Recommit CL5184 by bjornv@webrtc.org · 11 years ago
  60. 151cd25 Refactor Remote Estimators Test into a more reusable form. by solenberg@webrtc.org · 11 years ago
  61. 66d634f Revert 5184 "Small refactoring change in delay_estimator." by bjornv@webrtc.org · 11 years ago
  62. 2c75d4e Small refactoring change in delay_estimator. by bjornv@webrtc.org · 11 years ago
  63. 801822c Ensure that no packet stays in the pacer queue for longer than 2 seconds. by stefan@webrtc.org · 11 years ago
  64. 8f9da30 Create default implementation to fix issue in libjingle by sprang@webrtc.org · 11 years ago
  65. 4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
  66. 2622be1 Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. by asapersson@webrtc.org · 11 years ago
  67. 58b912b Remove const in vie_rtp_rtcp, where there is conflict with by sprang@webrtc.org · 11 years ago
  68. e7270f5 Faster implementation of BitRateStats. by mikhal@webrtc.org · 11 years ago
  69. 1a5aa03 Updated WebRTC version to 3.47 TBR=wu@webrtc.org by elham@webrtc.org · 11 years ago
  70. 5892ce5 Made video quality toolchain more configurable. by phoglund@webrtc.org · 11 years ago
  71. c86d1c6 Add include stdlib.h to files using abs. by stefan@webrtc.org · 11 years ago
  72. 586becf Add test for automatically disabling padding when no video is being captured. by stefan@webrtc.org · 11 years ago
  73. 5ae14be 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
  74. e8f79c5 Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. by turaj@webrtc.org · 11 years ago
  75. 8bdb87f Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. by sergeyu@chromium.org · 11 years ago
  76. 5fd393f Fix issues with sequence number wrap-around in jitter statistics. by turaj@webrtc.org · 11 years ago
  77. 6508af1 Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out. by turaj@webrtc.org · 11 years ago
  78. 44b21e7 Replace VideoFrameI420 with I420VideoFrame. by pbos@webrtc.org · 11 years ago
  79. 51fa6ac Don't reset the AEC filter in extended mode. by andrew@webrtc.org · 11 years ago
  80. ce4a0b8 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. by dwkang@webrtc.org · 11 years ago
  81. 970c5e5 Renaming ViEEncoderObserver::VideoSuspended by henrik.lundin@webrtc.org · 11 years ago
  82. a706baf Protect reads of ViEEncoder::video_suspended_. by pbos@webrtc.org · 11 years ago
  83. 9c15a62 Increase size of pacer window to 500 ms as that better matches the encoder. by stefan@webrtc.org · 11 years ago
  84. d7d60c8 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
  85. 0e6558b Lock access to ModuleRtpRtcpImpl::simulcast_. by pbos@webrtc.org · 11 years ago
  86. f8486d0 Rename DestroyStream methods to include Video. by pbos@webrtc.org · 11 years ago
  87. b87f528 Fix issues with sequence number wrap-around in jitter statistics by henrik.lundin@webrtc.org · 11 years ago
  88. 3c3a953 Add -Wnon-virtual-dtor warning for C++ code. by pbos@webrtc.org · 11 years ago
  89. e92aec9 Make interface destructor virtual by sprang@webrtc.org · 11 years ago
  90. 3fe2e7f Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
  91. 402f34c 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
  92. fa7ac56 Typo in vie_autotest_win.cc by braveyao@webrtc.org · 11 years ago
  93. 36fb531 Fixes a crash in VoE when unregistering JNI hooks. by henrike@webrtc.org · 11 years ago
  94. 13a4d31 Interface changes to old api, for use by new api transition. by sprang@webrtc.org · 11 years ago
  95. b06a926 Added ViE API for getting overuse measure. by asapersson@webrtc.org · 11 years ago
  96. 04bcc9d Deliver I420VideoFrames from VideoRender module. by pbos@webrtc.org · 11 years ago
  97. c2162d1 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
  98. f3b4602 Rename newapi::Transport::SendRTP()->SendRtp(). by pbos@webrtc.org · 11 years ago
  99. 60108c2 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
  100. 48cc9dc Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago